RE: [Asterisk-Users] Vlan question
Hi, > -Original Message- > I am setting up an Asterisk system with Cisco 7960 phones. I > have a PoE injector to insert between the patch panel and HP > 2626 switch. I plan to plug the users pc into the phone and > the phone into the wall. I would like the phones to have a > seperate subnet from the phones for performance reasons. > > May be a silly question, but with the pc and phone sharing > the same switch port, how will it know to seperate the > traffic and subnets? Simple answer: It can't. You can assign 1 VLAN or tell it to use more VLAN's and send the traffic tagged, but the switch will never know what packet belongs where. Apparently, Cisco 7960 can do some VLAN voodoo if you have them working with Cisco switches (proprietary stuff, fun eh). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP <->h.323
Hi Ryan! Interesting what experience you have made in this issue. We have setup the alternative channel for H.323 (the * built in chan_h323), and we are now in a testing phase. I was wondering (in case no transcoding is needed), how your setup treats the RTP streams. Do the RTP streams go end-to-end or always via Asterisk? Another question I'd be interested in: Have you also gained some experience with bridging _video_ calls between H.323 and SIP? cheers, Bernie PS: I'd be glad, if I also could get the relevant config files from you. On Fri, 13 Aug 2004, Ryan Wilkins wrote: > Yes, it can.. I'm doing it at my home. My current setup is > Asterisk-1.0-RC2 using the oh323 driver. I have a SIP connection to > Broadvoice talking to Asterisk. I have a e-tel (now Qtelnet) H.323 VoIP > telephone adapter as my end point talking to Asterisk. > > For processing sake, you may want to keep your codec the same all the way > through. Originally I ran G.711u on the SIP connection and G.711a on the > H.323 connection. It worked just fine but the logs always said something > about transcoding between u-law and a-law. I reset the H.323 link to > G.711u and now it says nothing about transcoding. In theory you would > lose a bit of audio quality in the translation process. In reality I > don't really know. > > email me privately if you want a sample config. > > Ryan Wilkins > > > On Fri, 13 Aug 2004, Yiannis Costopoulos, Web2Net Solutions Ltd. wrote: > > > is there a definite answer if asterisk can pass calls between SIP > > and h.323 protocols? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Q] DIDs
> Also known as DID service or called number information at various > times. You can have analog copper pots lines configured to send that > information. I don't know if Asterisk supports it. Anyone? This is called an Analog DID Trunk. Yes, Asterisk supports it. You can have inbound calls only on this type of line. Also, it does not support CallerID. The carrier's CO acts like a POTS analog handset. When a call comes in it simulates offhook status (puts a load on the line) and dials into your PBX (Asterisk), by sending DTMF. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Q] DIDs
On Sat, 14 Aug 2004, Robert Hajime Lanning wrote: > This is called an Analog DID Trunk. Yes, Asterisk supports it. > You can have inbound calls only on this type of line. Also, it > does not support CallerID. > > The carrier's CO acts like a POTS analog handset. When a call comes > in it simulates offhook status (puts a load on the line) and dials > into your PBX (Asterisk), by sending DTMF. I think it is possible to get DNIS+CallerID on a normal (bidirectional) analog line here in Sweden. We use DTMF CallerId delimited by the dtmf signals AB and CD. I don't have the reference for analog pstn connections at home so I cannot check if it was callerid+dnis or callerid+something else that could be passed. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te410p and Telstra Onramp 10
All set up and working now. My problem was that the cable from the E1 on the patch panel to the TE410p was too long. Also, there was no need to use DNIS, a standard entry matching the dialed number to the SIP extension was all that was necessary. Will play with using macros later once everything else has been bedded in. Craig - Original Message - From: "Craig Guy" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, August 14, 2004 10:21 AM Subject: Re: [Asterisk-Users] te410p and Telstra Onramp 10 > Thankyou very much for that :) I assume you would be using DNIS to match > the called number to internal extensions inside the dialplan? > > Our eval system used an X100p with internal Grandstreams. Call quality was > excellent except to mobile phones which had a pronounced echo that only the > internal party could hear. I am thinking this was due to the X100p and > should go away once using the onramp. I'll be putting in a wiki entry > regarding this config once it's all set up. > > Craig > > - Original Message - > From: "Adam Goryachev" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, August 14, 2004 9:36 AM > Subject: Re: [Asterisk-Users] te410p and Telstra Onramp 10 > > > > On Fri, 2004-08-13 at 22:06, Craig Guy wrote: > > > Hi, > > > > > > now that these cards have approval in Australia, has anyone had any luck > in > > > connecting them to a Telstra Onramp 10 service? Are these configured as > a > > > PRI, bchan=1-10, dchan=16 in zaptel.conf and switchtype=euroisdn in > > > zapata.conf? Better yet anyone have any of the above .conf files they > could > > > share? > > > > Yes, it works fine... although I get 'static' on calls connected to > > 'some' numbers (sometimes) Where static could be echo/feedback/weird > > sounds while the remote person talks. Only the remote person hears it. > > zaptel.conf: > > span=1,1,0,ccs,hdb3,crc4 > > bchan=1-10 > > unused=11-15,17-31 > > dchan=16 > > > > asterisk/zapata.conf > > [channels] > > usecallingpres=yes > > relaxdtmf=no > > rxgain=0.0 > > txgain=0.0 > > busydetect=no > > pridialplan=local > > nationalprefix=0 > > internationalprefix=0011 > > usecallerid=yes > > hidecallerid=no > > callwaiting=no > > callwaitingcallerid=no > > threewaycalling=no > > transfer=no > > cancallforward=no > > callreturn=no > > echocancel=yes > > echocancelwhenbridged=yes > > echotraining=yes > > callerid=asreceived > > adsi=no > > callprogress=no > > > > switchtype => euroisdn > > signalling => pri_cpe > > callgroup => 1 > > group => 2 > > immediate => no > > context => remote > > channel => 1-10 > > > > That's what I use... If anyone happens to see something I am doing which > > might break things/cause the problems I am having, please let me know... > > > > I would also be interested to hear how your setup goes... Mostly I have > > the problem when I call my mobile from a TDM FXS port on the same > > asterisk box. > > > > > Is an onramp 10 what is referred to as a 'channel bank'? > > > > No, channel bank converts the E1/T1 into analogue lines. > > > > Regards, > > Adam > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External MW Lamp On/Off
I have done something simmillar, but not the same. I send mwi notification to our softswitch (SIP). Basically I wrote a small app in pascal that sends a sip message to the softswitch. The app is called everytime a message is left or retrieved, using the extrennotify option in voicemail.conf. You could easily do something simillar, what you need to do, is write a script or app (if one does not already exist) that creates call file based on the parameters passed by externnotify. Hope this helps. Umar --- Greg Blakely <[EMAIL PROTECTED]> wrote: > One of the connections my asterisk PBX has is an > analog > extension from a Comdial hybrid. > > On the Comdial system, message waiting is turned on > by dialing > *3 and then the station number. > It is turned off by dialing #3 and the station > number. > > I was wanting to have Asterisk (or Comedian mail) > set the > message lamp in the Comdial system when a new > message arrives for a > user, and extinguish the lamp when the message has > been played. > > I understand that this has something to do with a > file that is > placed in /var/spool/asterisk/outgoing, but I have > no idea about > > + what the contents of that file should be, > + how Comedian mail would initiate putting the file > into the > outgoing queue, and > + how Comedian mail would initiate putting the > 'extinguish' file > into the outgoing queue. > > Has anyone done this sort of thing already? If so, > can you > point me in the right direction? > > As I mentioned in yesterday's post, I did find a > question and > partial answer to this in the asterisk-users > archives, but I need a bit > more information before I can make it work for me. > > Thanks in advance for any help you can give me. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
[ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] > > The problem I'm experiencing with many GS adapters, regardless of > firmware version is this. Call from one phone to another phone using > both the 'T' and 't' flags in the Dial() command. After they are > connected, you should be able to press '#' on either phone to hear > "transfer". What I am experiencing is the calling GS adapter will > hear "transfer" when they press '#', but when the receiving GS adapter > presses '#', nothing happens at all. Are you able to repeat this? If > not, can you please tell me the firmware revisions and Asterisk > version that you are using? > > Thank you very much. Yes, I am experiencing the same problem. Works fine with the BudgetTone phones but not with the HandyTone-286. Have you resolved this yet? We've been trying to figure this out on and off for the past couple of weeks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
Yes, we are experiencing the same problem and because of that we switched the called HT ata to Cisco ATA 186 ... Lubo Andy Lee wrote: [ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] The problem I'm experiencing with many GS adapters, regardless of firmware version is this. Call from one phone to another phone using both the 'T' and 't' flags in the Dial() command. After they are connected, you should be able to press '#' on either phone to hear "transfer". What I am experiencing is the calling GS adapter will hear "transfer" when they press '#', but when the receiving GS adapter presses '#', nothing happens at all. Are you able to repeat this? If not, can you please tell me the firmware revisions and Asterisk version that you are using? Thank you very much. Yes, I am experiencing the same problem. Works fine with the BudgetTone phones but not with the HandyTone-286. Have you resolved this yet? We've been trying to figure this out on and off for the past couple of weeks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Appradius Project: RADIUS authentication and accounting support for Asterisk PBX http://appradius.minitelecom.org/ - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Howto remove digits from a called number
Hi list,
I have SIP clients and H323 GK connected through h323 channel (Nufone).
In h323 conf I gave prefix=09 so all call starting with this prefix are
send to asterisk. The context is also given their as [fromh323]
But now, in asterisk, I want to have the called number without this 2
leading digits so the exten variable will be according to my actual
dialplan. Here's an exemple:
In extensions.conf I have
exten => 100,1,Goto(demo,s,1)
If I call #100 from SIP it's ok. So now, if I want to reach this
extension from an h323 EP, I have to call 09100. This call will never
succeed (or I create a new exten line, same as above, with this prefix).
Till now I had prefix=1,2,3,4,5,6,7,8,9 in my h323.conf, so all call not
starting with 0 are directed to *, others to a Gateway. In this case
it's working, but endpoints are not able to call IP adresses :-(, those
call are redirected to *
I tried with macro but a line like
${ARG1} => 100,1,Goto(demo,s,1)
is not accepted (no such host)
Is their not a variable containing the dialed number that could be
modified? Is it not recommended? Has someone another solution (yes I
know, I can duplicate my lines, but that's not what I want ;-))?
Thanks for any hint
--
Daniel
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Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
Unfortunately the Cisco ATA-186 does not support iLBC which means extra costs for purchasing 729 licenses. The ATA-286 works fine other than this 1 issue. Do the Grandstream developers follow this list?? This problem has been persistent for a LONG time and each new firmware version still has it unfixed!! On Sat, 14 Aug 2004 19:36:43 +0300, Lubomir Christov <[EMAIL PROTECTED]> wrote: > > Yes, we are experiencing the same problem and because of that we > switched the called HT ata to Cisco ATA 186 ... > > Lubo > > > > Andy Lee wrote: > > [ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] > > > >>The problem I'm experiencing with many GS adapters, regardless of > >>firmware version is this. Call from one phone to another phone using > >>both the 'T' and 't' flags in the Dial() command. After they are > >>connected, you should be able to press '#' on either phone to hear > >>"transfer". What I am experiencing is the calling GS adapter will > >>hear "transfer" when they press '#', but when the receiving GS adapter > >>presses '#', nothing happens at all. Are you able to repeat this? If > >>not, can you please tell me the firmware revisions and Asterisk > >>version that you are using? > >> > >>Thank you very much. > > > > > > Yes, I am experiencing the same problem. Works fine with the > > BudgetTone phones but not with the HandyTone-286. > > > > Have you resolved this yet? We've been trying to figure this out on > > and off for the past couple of weeks. > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > - > Appradius Project: RADIUS authentication and accounting support for > Asterisk PBX > http://appradius.minitelecom.org/ > - > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto remove digits from a called number
On Sat, 14 Aug 2004, administrator tootai wrote:
> Hi list,
>
> I have SIP clients and H323 GK connected through h323 channel (Nufone).
> In h323 conf I gave prefix=09 so all call starting with this prefix are
> send to asterisk. The context is also given their as [fromh323]
>
> But now, in asterisk, I want to have the called number without this 2
> leading digits so the exten variable will be according to my actual
> dialplan. Here's an exemple:
>
> In extensions.conf I have
>
> exten => 100,1,Goto(demo,s,1)
>
> If I call #100 from SIP it's ok. So now, if I want to reach this
> extension from an h323 EP, I have to call 09100. This call will never
> succeed (or I create a new exten line, same as above, with this prefix).
You're right, you will have to create an extension to match the 09xxx
numbers. But you don't have to create one for every "real" SIP extension
you have. Instead, make one that matches all 09xxx extensions and does a
goto:
exten => _09XXX,1,Goto(yoursipextensionscontext,${EXTEN:2},1)
for three digit "real" extensions. Add or remove X's for more or fewer
digits, or just use _09. for _ANYTHING_ that starts with 09 (keep that in
mind.. sometimes that wildcard extension comes back to bite you!).
Greg
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Re: [Asterisk-Users] Confused --> Hardware specs
Are you going to use single E1 line? How many concurrent calls. Single Xeon 3.0 with *1 GB *ram should serve 30-32 calls with not problem, even during G729 transcoding and echo cancellation. If you are not plaining to do G729 transcoding - then I believe you can put more then 32 calls in this box. Having kept an Eye on hardware specs mails in the list, and having read the hardware notes on voip-info, I am still confused over the recommended hardware: We are going to have up to 100 operators and staff making calls, mostly through zap connected to TE405P through to a 32-channel E1. However, it is possible that we may need another E1 - what is the rough ratio between operators and calls when the call length on average is 3 minutes?) I was thinking of adding some VOIP into the mix. So, what speed proc should I use ? I am currently looking at: A) Dual-Xeon 2.8 with 2 GB ram and 36 gb raid-1 B) Single Xeon 3.0 with 2 GB ram and 36 gb raid-1 So, assuming I stick with 1 E1 line, is a dual better than a single, or should I take the savings from buying a single Xeon, and buy two machines with two TE405p cards ? Help! Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External MW Lamp On/Off
Yes, it helps quite a bit. It shows me where Comedian Mail spawns the external app. Do you have a copy of your SIP MWI script? I may be able to use it as a starting point. Also, can you tell me what variables are passed from asterisk to the app? Thank you very much. Greg > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear > Sent: Saturday, August 14, 2004 7:10 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] External MW Lamp On/Off > > I have done something simmillar, but not the same. > > I send mwi notification to our softswitch (SIP). > Basically I wrote a small app in pascal that sends a sip > message to the softswitch. The app is called everytime a > message is left or retrieved, using the extrennotify option > in voicemail.conf. > > You could easily do something simillar, what you need to do, > is write a script or app (if one does not already exist) that > creates call file based on the parameters passed by externnotify. > > Hope this helps. > > Umar > --- Greg Blakely <[EMAIL PROTECTED]> wrote: > > One of the connections my asterisk PBX has is an analog > extension > > from a Comdial hybrid. > > > > On the Comdial system, message waiting is turned on by dialing > > *3 and then the station number. > > It is turned off by dialing #3 and the station number. > > > > I was wanting to have Asterisk (or Comedian mail) set > the message > > lamp in the Comdial system when a new message arrives for a > user, and > > extinguish the lamp when the message has been played. > > > > I understand that this has something to do with a file > that is placed > > in /var/spool/asterisk/outgoing, but I have no idea about > > > > + what the contents of that file should be, > > + how Comedian mail would initiate putting the file > into the outgoing > > queue, and > > + how Comedian mail would initiate putting the > 'extinguish' file into > > the outgoing queue. > > > > Has anyone done this sort of thing already? If so, can > you point me > > in the right direction? > > > > As I mentioned in yesterday's post, I did find a > question and partial > > answer to this in the asterisk-users archives, but I need a > bit more > > information before I can make it work for me. > > > > Thanks in advance for any help you can give me. > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ALL > -NEW Yahoo! Messenger - all new features - even more fun! > http://uk.messenger.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Questions
Hi, I'm new to Asterisk and there are several questions on it: 1. does it support (or will it support) Fax in the neareast future ? 2. will be Windows version developed ? 3. Are there any known addons that allow to use Asterisk as: IP-centrex system call center solution complete calling card solution Ildar
[Asterisk-Users] Re: Questions
Please don't post in HTML; use plain text instead. Ildar Gabdulline <[EMAIL PROTECTED]> wrote: > > I'm new to Asterisk and there are several questions on it: > 1. does it support (or will it support) Fax in the neareast future ? Yes, but you need to add a component called spandsp. > 2. will be Windows version developed ? No. Why would anyone want to? Linux is far superior for this kind of app. > 3. Are there any known addons that allow to use Asterisk as: > IP-centrex system > call center solution > complete calling card solution All these things are possible, but you will need to learn a lot. Asterisk has a *lot* of power, but takes time and effort to learn. You will find a huge amount of information on the Wiki, which is at http://www.voip-info.org Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help - is voip good for in-house calls?
Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and standards as a PBX-only, with "traditional" phones? Or should I go all the way to Digium's TDM? Or should I forget the whole thing and get a traditional PBX? ;) My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. I've read lots about voip, and I'm quite impressed with it, but most case studies show voip being used to interconnect offices. My case is different - I want to replace a traditional PBX to handle in-house phone calls, either from room-to-room in the same building and room-to-POTS. Any comment, help, tip or link would be greatly appreciated! Yours truly, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List traffic/Software
I know this has been gone over before but It seems that most of the traffic to this list is the same 10 questions being asked over and over again. What if someone (with some programing skill) wrote a script so that when someone posted to the list, it would search the wiki and google and respond to them. If they where not happy with the response, then they could forward there message back to the list and it would go out to the 8000 people. While there are lots of issues with doing this, I do think it would cut down on the amount of "newbie" traffic that this list gets. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Asterisk should work fine for this application - but you and/or your users may be expecting the Grandstreams to look/act like traditional key system phones, where you've got a bunch of buttons labeled "Computer Room" or "Joe" and "Bob", or whatever, where you can press that button to call that extension. The Budgetones don't do that - users will need to remember (or have an extension list to tell them) that the Computer Room is at extension 110, and Joe is at 111, and Bob is 112, and so forth. My suggestion is that you buy 2 Budgetones and set up Asterisk on an old PC - so your total investment in the experiment will be < $200. Get that up & running, and let users play with the phones and the functions you can provide. If they like it, great. If they don't like it, you're not out much money, and you ought to be able to resell the Budgetones for something like 80% of the new price on Ebay or whatever. You can get set up with incoming and outgoing IAX connections via someone like Voicepulse or Nufone or IPKall (or some combination thereof) so you can even let people experiment with incoming and outgoing call quality and behavior without spending a lot on interface cards. You might also look at some of the other VoIP phones, which aren't a whole lot more money and might look more like the PBX/key phones that people are used to. The Budgetones are more similar to consumer/home telephones from the early 1990's. -- Greg Broiles, JD, EA [EMAIL PROTECTED] (Lists only. Not for confidential communications.) Law Office of Gregory A. Broiles San Jose, CA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help - is voip good for in-house calls?
Hello Francis, My office build is the same as yours. 15 or so extensions, low traffic 100MB network, and a desire to have a phone system that uses VoIP. I have my system working as a PBX just like you would. I use two TDM400s for my 8 POTS lines and Polycom IP 500 phones at the desktop. I also tested with the Grandstream phones you suggested. SO, we have the same system requirements so here are the answers as I have found them for my implementation Voice quality on the SIP based phones has a lot to do with the codec you use. The lowest compression codec is uLaw and that is what I use since we have tons of bandwidth to spare. Also, my HP switch has COS (class of service which is like QOS) so I can prioritize the packets coming from my phones over the standard network traffic. Even without this switching feature turned on, performance was great. The phones themselves play another role in the quality. Grandstreams are pretty good and I have only used mine for testing so I will not disparage them. However, the old saying stands. You get what you pay for. Raising your phone budget from $85 to more like $150-250 will get you a phone with more features and greater expandability in my IHO. However, you can still do great things with the cheaper Grandstream phones and still have a system that works very well. IT is all up to what you can spend and what you need. Google the archive by putting "site:lists.digium.com" in front of your search string (no quotes though). You should see some good info on phones. Latency is gonna be there on any network. However, on my network (which is just like yours) the latency is very very low. We are talking 20-40ms tops and it is completely unnoticeable when using the phone. The only problem I have had at all has been with occasional echo. It does not happen often and it usually takes about 5 seconds for the * box to train up and remove it. Most of this seems to originate in the fact that I am using POTS lines. The solution that uses a T1 PRI has better features and I think it has less echo potential. However, that would not work for me since my T1 provider wanted to make me pay 6 grand to switch to a PRI from my standard data T1 with POTS. Just some food for thought... I have been a VoIP user for about 1 month after spending another researching what when where how... So, we know I am not an expert... but as a fellow user and new VoIP initiate, I can tell you that Asterisk is a phenomenal product for SMB level offices like yours and mine. When I compared it to a PBX system of comparable power, expandability, and feature set, Asterisk won easily since the only real cost I have had was for my phones. I have my system in place for around 3000 dollars and it is competitive with all the 10K dollar solutions the vendors threw at me plus it has an undeniable advantage in upgrade path. All upgrades to the system are free and the sky is the limit to what you can build using the framework that all the * gurus have built into this system. Not to mention the fact that if anything ever goes wrong with the server, I can have a new one in place in under and hour. Try that with a PBX when some proprietary part goes belly up. You could wait days potentially. My $.02. Hope this helps. Cheers, Wiley -Original Message- From: Francis Augusto Medeiros [mailto:[EMAIL PROTECTED] Sent: Saturday, August 14, 2004 1:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help - is voip good for in-house calls? Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and standards as a PBX-only, with "traditional" phones? Or should I go all the way to Digium's TDM? Or should I forget the whole thing and get a traditional PBX? ;) My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. I've read lots about voip, and I'm quite impressed with it, but most case studies show voip being used to interconnect offices. My case is different - I want to replace a traditional PBX to handle in-house phone calls, either from room-to-room in the same building and room-to-POTS. Any comment, help, tip or link would be greatly appreciated! Yours truly, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options v
Re: [Asterisk-Users] BugetTone Bug Showstopper,
Use the 'send' button On Jul 29, 2004, at 3:26 PM, Kanuri, Seshu wrote: I have setup Grandstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Foster Sent: Thursday, July 29, 2004 9:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BugetTone Bug Showstopper, On Wed, 28 Jul 2004 23:31:06 -0400, Seth Remington <[EMAIL PROTECTED]> wrote: On Wed, 2004-07-28 at 21:00, James Gardiner wrote: How do I get Asterisk to recognise the # key from the granstream phone for doing transfers? Make sure the Grandstream is configured to send DTMF via SIP INFO instead of in-audio. -Seth Also, don't forget to disable the "#-key as redial" feature. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 shared lines
No luck on that here either, but the latest sip and firmware updates have added xml mini browser capability to the IP600 phones. That means you should be able to add xml code like this: http://cvs.largeone.net/index.cgi/*checkout*/asterisk/scripts/status.cgi and get the status of other lines. John Chris HARIGA wrote: Hi, We have some Polycom IP600 phones and I try to setup "shared" extensions to see the status of the line and didn't work. I would like to know if someone has this feature on the phone using Asterisk. Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Dear Greg, Thanks a lot for your e-mail! Here are my comments: On Sat, 14 Aug 2004 14:37:08 -0700, Greg Broiles <[EMAIL PROTECTED]> wrote: > Asterisk should work fine for this application - but you and/or your > users may be expecting the Grandstreams to look/act like traditional > key system phones, where you've got a bunch of buttons labeled > "Computer Room" or "Joe" and "Bob", or whatever, where you can press > that button to call that extension. > > The Budgetones don't do that - users will need to remember (or have an > extension list to tell them) that the Computer Room is at extension > 110, and Joe is at 111, and Bob is 112, and so forth. This is no problem, as with the old PBX we also had to do that, and without LCD. We used a small PBX with regular phones. My concern was if I'd have to teach folks how to dial, but I guess that I can still have the option to assign a number that will give immediate access to the PSTN, so no need to make a special dialplan to acomodate the weird numbering system we have in Brazil (sometimes we dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) > My suggestion is that you buy 2 Budgetones and set up Asterisk on an > old PC - so your total investment in the experiment will be < $200. > Get that up & running, and let users play with the phones and the > functions you can provide. If they like it, great. If they don't like > it, you're not out much money, and you ought to be able to resell the > Budgetones for something like 80% of the new price on Ebay or > whatever. You can get set up with incoming and outgoing IAX > connections via someone like Voicepulse or Nufone or IPKall (or some > combination thereof) so you can even let people experiment with > incoming and outgoing call quality and behavior without spending a lot > on interface cards. This is really a great idea. See, my biggest concern is not the voice quality in terms of audio, but if the conversation is allowed to flow in the same way as with regular phones. Or, in other words, if there are significant delays that makes the communication a bit frustrating. Brazilians do interrupt each other a lot while talking on the phone and, on voip calls over the internet with slow connections, this habit made the conversation a bit weird, as sometimes we couldn't realize if the other part started to talk again... :) We don't have in Brazil, as far as I know, voice providers such as those you suggested. So I'll definetly need an FXO. I'm also considering a media gateway - I don't know really if they work the same way for the end user. > You might also look at some of the other VoIP phones, which aren't a > whole lot more money and might look more like the PBX/key phones that > people are used to. The Budgetones are more similar to consumer/home > telephones from the early 1990's. I haven't find any other phone below $80. Budgettones, if they work good, seems to be the best option for us. Again, thanks a lot for helping!! Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Hi there Wiley! On Sat, 14 Aug 2004 14:43:05 -0700, Wiley E. Siler <[EMAIL PROTECTED]> wrote: > My office build is the same as yours. 15 or so extensions, low traffic > 100MB network, and a desire to have a phone system that uses VoIP. I > have my system working as a PBX just like you would. I use two TDM400s > for my 8 POTS lines and Polycom IP 500 phones at the desktop. I also > tested with the Grandstream phones you suggested. SO, we have the same > system requirements so here are the answers as I have found them for my > implementation Thanks for your e-mail!!! Your setup and your envoironment are really encouraging, since they are very similar to what I have in mind (except for the quantity of POTS lines - we won't use that many). > Voice quality on the SIP based phones has a lot to do with the codec you > use. The lowest compression codec is uLaw and that is what I use since > we have tons of bandwidth to spare. Also, my HP switch has COS (class > of service which is like QOS) so I can prioritize the packets coming > from my phones over the standard network traffic. Even without this > switching feature turned on, performance was great. The phones > themselves play another role in the quality. Grandstreams are pretty > good and I have only used mine for testing so I will not disparage them. > However, the old saying stands. You get what you pay for. Raising your > phone budget from $85 to more like $150-250 will get you a phone with > more features and greater expandability in my IHO. However, you can > still do great things with the cheaper Grandstream phones and still have > a system that works very well. IT is all up to what you can spend and > what you need. Google the archive by putting "site:lists.digium.com" in > front of your search string (no quotes though). You should see some > good info on phones. Well, I'm not really looking for a lot of phone features, just the basics (transfers, call retrieval, etc.). And voice quality is not what I am most worried about, but the delays on the conversation. However, on your mail, you say that latency is, in most cases, unnoticeable, and those are great news to me, as I feel more comfortable to suggest our office to buy ip-phones and use them, knowing they will serve us well. > Latency is gonna be there on any network. However, on my network (which > is just like yours) the latency is very very low. We are talking > 20-40ms tops and it is completely unnoticeable when using the phone. > The only problem I have had at all has been with occasional echo. It > does not happen often and it usually takes about 5 seconds for the * box > to train up and remove it. Most of this seems to originate in the fact > that I am using POTS lines. The solution that uses a T1 PRI has better > features and I think it has less echo potential. However, that would > not work for me since my T1 provider wanted to make me pay 6 grand to > switch to a PRI from my standard data T1 with POTS. Just some food for > thought... I'll most likely use a BRI. Do you think this will help to avoid echo? > I have been a VoIP user for about 1 month after spending another > researching what when where how... So, we know I am not an expert... > but as a fellow user and new VoIP initiate, I can tell you that Asterisk > is a phenomenal product for SMB level offices like yours and mine. When > I compared it to a PBX system of comparable power, expandability, and > feature set, Asterisk won easily since the only real cost I have had was > for my phones. I have my system in place for around 3000 dollars and it > is competitive with all the 10K dollar solutions the vendors threw at me > plus it has an undeniable advantage in upgrade path. All upgrades to > the system are free and the sky is the limit to what you can build using > the framework that all the * gurus have built into this system. Not to > mention the fact that if anything ever goes wrong with the server, I can > have a new one in place in under and hour. Try that with a PBX when > some proprietary part goes belly up. You could wait days potentially. > My $.02. Hope this helps. That's also what I hope it will happen here! If we want to expand, we don't want to end up with a closed-system that won't handle more extensions or phone lines. And since things are converging, and things like FWD, Vonage and others are helping ppl to communicate, the use of a voip based system would certainly help us more to communicate with our clients and with ourselves. Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Installing Zaptel Modules on Fedora Core 2
I read a few discussions on installing Zaptel modules in Fedora Core 2 with 2.6.5 kernel. I was wondering if there is a definitive FAQ on this? I am still unable to install by FXO card in my pbx box because the modules won't install. Thanks Vikas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: > My concern was if I'd have to teach folks how to dial, but I guess > that I can still have the option to assign a number that will give > immediate access to the PSTN, so no need to make a special dialplan to > acomodate the weird numbering system we have in Brazil (sometimes we > dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) With overlap dialing enabled in Asterisk this should work. We have a similar setup. Note that Asterisk is mostly tested with enbloc-dialing which seems to be the norm in USA where the numbering plan has a fixed length. We have gotten overlap dialing to work correctly except for the call records but they are not that important to us. I hope to create a patch for that after 1.0 has been released. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help - is voip good for in-house calls?
Hello Francis, > I'll most likely use a BRI. Do you think this will help to avoid echo? I could not say as I have never used a BRI and I am pretty new to this too. I do know that BRI is supported from watching conversations in this email list and reading online. People seem to use it a bit so it must work well. Googling the list with BRI should get you tons of good leads. Greg had a great idea in having you set it up and try it. In fact, that is exactly how I did mine. I purchase a cheap clone card for $15 and used it to test on one POTS line while I tweaked my configuration files and got the system validated. I tested the system with soft phones, one Polycom IP 500, and one Grandstream Budgetone 101. The Budgetone worked well and was leagues easier to setup than my Polycom actually. For expandability, I believe that the cap I have seen is about 60 concurrent calls for one Asterisk box and that is with a pretty serious server by most users standards. I cannot imagine having that many calls at this point so I am fine but I jus though t you would want to know. The nice thing about * is that you can just build another server and link them together over IAX. Again, the low cost of implementation pays off and you get to continue growth. I will never go back to proprietary PBX now that I finally have a solution that I can control. Cheers, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: > I'll most likely use a BRI. Do you think this will help to avoid echo? Using a BRI will eliminate echos from the pstn connection. Your ip phones should prevent echos from the local phone connections as well. That way you should not cause any noticable echo for the remote party. Being all digital has its advantages. :-) Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help - is voip good for in-house calls?
On Sat, 14 Aug 2004, Wiley E. Siler wrote: > Greg had a great idea in having you set it up and try it. In fact, that > is exactly how I did mine. I purchase a cheap clone card for $15 and > used it to test on one POTS line while I tweaked my configuration files > and got the system validated. I tested the system with soft phones, one > Polycom IP 500, and one Grandstream Budgetone 101. The Budgetone worked > well and was leagues easier to setup than my Polycom actually. Using a pots may not give an accurate picture. It is a source of echos which can, when combined with a slight latency introduced in the voip links, change an acceptable reverb to a nasty echo. The cheap clone cards are all of the x100 card I believe. It has a fixed impedance of 600 ohms pure resistive. A lot of countries outside USA seem to use other line impedances. The mismatch leads to echos. > For expandability, I believe that the cap I have seen is about 60 > concurrent calls for one Asterisk box and that is with a pretty serious > server by most users standards. I cannot imagine having that many calls > at this point so I am fine but I jus though t you would want to know. > The nice thing about * is that you can just build another server and > link them together over IAX. Again, the low cost of implementation pays > off and you get to continue growth. I will never go back to proprietary > PBX now that I finally have a solution that I can control. With no transcoding you should be able to go higher than that I expect. For local ip phones g.711 is probably usable. Most low numbers I have seen seemed to do a lot of transcoding to liwer bitrate formats. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 00:22:42 +0200 (CEST), Peter Svensson <[EMAIL PROTECTED]> wrote: > On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: > > > My concern was if I'd have to teach folks how to dial, but I guess > > that I can still have the option to assign a number that will give > > immediate access to the PSTN, so no need to make a special dialplan to > > acomodate the weird numbering system we have in Brazil (sometimes we > > dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) > > With overlap dialing enabled in Asterisk this should work. We have a > similar setup. Note that Asterisk is mostly tested with enbloc-dialing > which seems to be the norm in USA where the numbering plan has a fixed > length. We have gotten overlap dialing to work correctly except for the > call records but they are not that important to us. I hope to create a > patch for that after 1.0 has been released. > > Peter Hej Peter, Hmmm... I'll have to read more about overlap dialing - haven't noticed it while reading the docs. Thanks, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 00:29:21 +0200 (CEST), Peter Svensson <[EMAIL PROTECTED]> wrote: > On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: > > > I'll most likely use a BRI. Do you think this will help to avoid echo? > > Using a BRI will eliminate echos from the pstn connection. Your ip phones > should prevent echos from the local phone connections as well. That way > you should not cause any noticable echo for the remote party. Being all > digital has its advantages. :-) GREAT news! :)) My uncle's has an old Teles.ISDN card hanging up useless in his computer... :) Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Saturday 14 August 2004 18:29, Peter Svensson wrote: > Using a BRI will eliminate echos from the pstn connection. Your ip phones > should prevent echos from the local phone connections as well. That way > you should not cause any noticable echo for the remote party. Being all > digital has its advantages. :-) *wrong*, unless there is something drastically different between ISDN PRI and ISDN BRI. I have *bad* echo on my Bell Canada PRI (via TE405P) -- software echo cancellation helps but it's not perfect. All the BRI/PRI does is ensure that *your end* is not generating echo, as there is no hybrid there. There is still a hybrid in your handset, and there is still a hybrid at the other end of the PSTN call. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free MOH MP3
Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone’s post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler
RE: [Asterisk-Users] Free MOH MP3
The wiki is your friend, found it in under 30 seconds. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold Under also see: * Sounddogs http://www.sounddogs.com/catsearch.asp?Type=2 Royalty Free Music * FreeMusic http://hebb.mit.edu/FreeMusic/ Free Classical Music From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler Sent: Saturday, August 14, 2004 7:51 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Free MOH MP3 Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Free MOH MP3
Well, yes it is. Sorry about that. I didn't even think about the Wiki since what I was looking for was content. I just googled against the list thinking that was where I saw it. Thanks! Cheers, Wiley > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Saturday, August 14, 2004 5:01 PM > To: 'Bill Church'; [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Free MOH MP3 > > The wiki is your friend, found it in under 30 seconds. > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHol d > > Under also see: > > * Sounddogs http://www.sounddogs.com/catsearch.asp?Type=2 Royalty Free > Music > * FreeMusic http://hebb.mit.edu/FreeMusic/ Free Classical Music > > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler > Sent: Saturday, August 14, 2004 7:51 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Free MOH MP3 > > > > Hello All, > > > > Sorry to rehash a question I am sure has shown several time but I cannot > google up the answer from the lists. > > > > Does anyone know where I can get some royalty free, cost free music for my > music on hold? > > > > I saw someone's post several weeks ago that said that this exists at a > download site but I have not been able to find it. > > > > Thanks! > > Wiley Siler > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP600 shared lines
Hi, Thanks for your reply. I will try the script... Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Saturday, August 14, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP600 shared lines No luck on that here either, but the latest sip and firmware updates have added xml mini browser capability to the IP600 phones. That means you should be able to add xml code like this: http://cvs.largeone.net/index.cgi/*checkout*/asterisk/scripts/status.cgi and get the status of other lines. John Chris HARIGA wrote: > Hi, > > We have some Polycom IP600 phones and I try to setup "shared" extensions to > see the status of the line and didn't work. > I would like to know if someone has this feature on the phone using > Asterisk. > > Best regards, > > Chris HARIGA > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Questions on various and sundry IP phones, and cabling
I'm attempting to do a first-time Asterisk install at home, firstly for use by my self and my family, and secondly as a learning experience. I've got a new house, and the previous owners removed all but one (1) phone jack. So I figured I might as well build a PBX. Functional goals include station-to-station calling, rudimentary auto attendant/voice mail, and perhaps tieing into the Altigen box that I've got at work via h.323. But my first goal is to get any of my devices to talk to Asterisk, which so far I've been unable to do. Hardware in-hand consists of a K6-2 based 2.6 kernel Gentoo box, two Selsius 30 VIP phones, a couple of Selsius 12SP+'s, a loaner Altigen IP-600 (which is nothing like a Polycom IP600, despite the name), and a switched 100 megabit network. There's an X100P on order which should be here real soon now, which will be sharing the house's solitary CO line with a Uniden cordless phone until I get an FXS interface of some kind (which won't be until after the rest of the kit all works). I've been reading list archives and the wiki pages for weeks, but now that I'm starting actual implementation, there's a few things that I'm not sure how to go about. So. The questions: 1. Is the K6-2 meaty enough to do function in this enviroment? I don't ever anticipate doing any hefty compression, as this is all in-house (unless I get the remote Altigen box flying, which will add one potential g.723 connection...). Routing calls around between skinny phones and a single PSTN connection shouldn't be very taxing, should it? 2. Asterisk seems to support the Selsius phones, maybe, via chan_sccp or chan_skinny. I say "maybe" because, though some people on this list seem to have them working fine, documentation on any of it is extremely lacking. The phones are -old-, booting to a 1997 copyright screen and reporting K2.01 as a version. The results so far seem somewhat atypical of what's reported on this list. Currently, I've got a 30 VIP configured with DHCP enabled, and have the TFTP server configured with Aterisk's address. The phone boots, attempts to request SEPDefault.cnf and SEP0010EB001BCA.cnf (neither of which exist) via TFTP, then reboots a few minutes later and tries again. Asterisk console with a dozen -v's and a -d doesn't show any activity during this time, whether using chan_skinny or chan_sccp. Behavior is the same with recent Asterisk CVS or the 0.9.0 ebuild in portage. Standard behavior, as far as I can tell, is to at least see a flurry of registration attempts. Is there any way to better see what all is going on? It's important that these phones work, because without them, there's no point in going any further with the project. They were free. :) ISTR some information on Selsius phones specifically, lurking at Lambda Solutions, but it's all gone now. 3. The IP-600. For those unfamiliar, it's more-or-less a generic h.323 endpoint, with the usual settings for gatekeeper and such, along with some Altigen extensions that are safe to ignore. It does not speak SIP, or any other protocol. In its native Altigen enviroment, it will only speak either g.711 and g.723.1 6.3k, but the phone itself might support additional codecs. I've successfully used this phone by itself to call Netmeeting without a hitch. Currently, my Asterisk build lacks h.323 support, because neither the builtin h323 channel nor oh323 are willing to compile with CVS (and gentoo's non-cvs zaptel driver is unhappy with kernel 2.6), but I'll probably be able to make that end of things work on my own. When I finally do get the an h.323 channel configured, what sort of behavior should I expect out of a default Asterisk install? Should I be able to ring the server's IP and get back a generic auto attendant, or what? In other words, how much configuring of Asterisk is necessary to see that h323 (or skinny, for that matter) is working between the phone and Asterisk and get audio going in both directions? And, again, is there a better way to see what's going on than the messages in the Asterisk console? 4. As I mentioned earlier, for the time being, Asterisk will be sharing it's phone line with a cordless phone. Is the combination of the X100P card and Asterisk enough to detect when the cordless is off-hook, and speak an error message instead of stupidly attempting to dial out? I intend to have Asterisk answer all incoming calls just after CID reception, and present an auto-attendant/telemarketer filter, with an option to immediately drop the call. The desired behavior is thus as follows: Inbound: Asterisk answers after two rings and presents an auto attendant. I elect to answer the call with the (non-*) cordless, hit (say) 8, and Asterisk hangs up and gets out of the way. Else, keep the attendant going and act like a PBX. Outbound: When connecting to the PSTN, Asterisk first checks to see if the line is in use (by sensing loop voltag
Re: [Asterisk-Users] Analog Phones with Status Light Indicators
> For those that have been selling telecomm for awhile, its fairly well known > the business purchasing decision is based primarily on "cost" followed by > "features". Its also fairly well understood that many businesses will > list a feature or two as "required" to ensure their favorite vendor is > selected. Asterisk can be sold into some of those accounts but not all. Thanks for taking the time to explain that to me (and hopefully a number of the other people who have been repeating the same questions)... Also, specifically, thanks for quoting the last paragraph. I happen to be the 'preferred' supplier, and I'm reasonably confident that I can provide the solution at a lower cost with more features than the competitor. So, I'll go now and write up my proposal, and see where I end up. I am wondering when those open-source IAX based phones are going to be released, and I wish we still had a IAX load for the snom 200 phones. Either one of those would have to benefit the community in this as well as other features. Thanks, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List traffic/Software
I'm a noob yes, however I usually read all the docs before asking dumb question. Perhaps what is needed is better documentation? :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, August 14, 2004 2:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] List traffic/Software I know this has been gone over before but It seems that most of the traffic to this list is the same 10 questions being asked over and over again. What if someone (with some programing skill) wrote a script so that when someone posted to the list, it would search the wiki and google and respond to them. If they where not happy with the response, then they could forward there message back to the list and it would go out to the 8000 people. While there are lots of issues with doing this, I do think it would cut down on the amount of "newbie" traffic that this list gets. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 shared lines
Lots of luck with that one. You may want to wait until I get some specs on the xml from my vendor. I'll make sure they get to the wiki. Heck, I may even try writing some code to do the trick for both of us. In the meantime, if you're pressed for time, you may want to try a SER - asterisk combo. John Chris HARIGA wrote: Hi, Thanks for your reply. I will try the script... Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Saturday, August 14, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP600 shared lines No luck on that here either, but the latest sip and firmware updates have added xml mini browser capability to the IP600 phones. That means you should be able to add xml code like this: http://cvs.largeone.net/index.cgi/*checkout*/asterisk/scripts/status.cgi and get the status of other lines. John Chris HARIGA wrote: Hi, We have some Polycom IP600 phones and I try to setup "shared" extensions to see the status of the line and didn't work. I would like to know if someone has this feature on the phone using Asterisk. Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pthread issue
I hope its okay to post such a long question, but thanks in advance for reading it I have been trying to get asterisk to run, I have played with it. and searched on the wiki and list to no avail asterisk compiles succesfully ( yes I installed all dependenys ) I am running gentoo 2004.2 with a 2.4 kernel but when I try to run asterisk I get ( last 5 lines of output from -vvvc ) == Registered channel type 'vpb' (Standard VoiceTronix API Driver) [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == Starting vpb monitor thread[81925] Segmentation fault VoIP root # output with -vvvgc [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found Aug 14 13:12:49 266 WARNING[16384]: chan_mgcp.c:4032 reload_config: Unable to get our IP address, MGCP disabled == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) Segmentation fault VoIP root # so I decided to use gdb >> gd(gdb) set args -vvvgc >> (gdb) run Starting program: /usr/sbin/asterisk -vvvgc warning: Unable to find dynamic linker breakpoint function. GDB will be unable to debug shared library initializers and track explicitly loaded dynamic code. Program received signal SIG32, Real-time event 32. 0x400239a4 in pthread_getconcurrency () from /lib/libpthread.so.0 >> (gdb) bt #0 0x400239a4 in pthread_getconcurrency () from /lib/libpthread.so.0 #1 0x400237e8 in pthread_getconcurrency () from /lib/libpthread.so.0 #2 0x40022ea2 in pthread_create () from /lib/libpthread.so.0 #3 0x080aab97 in test_for_thread_safety () at utils.c:200 #4 0x08098bbc in main (argc=0, argv=0x1) at asterisk.c:1670 >> (gdb) quit >> The program is running. Exit anyway? (y or n) y VoIP root # so I figure something wrong with pthreads but thats where I run into a wall.. any ideas are most apreciated thanks in advance John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 help
I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 help
hi man, if you are trying to upgrade to the latest version, change the permissions of the file, then to the SIP.cnf file add a line that says image version = , copy that line from the Sipdefault.cnf file, . If the first workaround does not work, try to downgrade to version 2.3 and the do the upgrade directly from that version. I can provide you any image you need. Let me know how that works I will highly appreciate your answerJason Kawakami <[EMAIL PROTECTED]> wrote: I have 4 7960's that I am trying to get working but 2 of them will notupdate to the SIP image on my tftp server like the first ones did.i keep getting the error on the phone 'Defaulting CM to TFTP server' like itisn't seeing the *.bin on the server.are you supposed to have on of those for each phone? would be like cisco etal to do something like that.TIAJason Kawakami___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!
Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
For blind transfers, press flash twice, then press #. For consultative transfers, press flash once, talk to the other party and tells him to hangup, press flash again, then press #. - Original Message - From: "Dennis Cartier" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, August 15, 2004 2:10 AM Subject: Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm? > Unfortunately the Cisco ATA-186 does not support iLBC which means > extra costs for purchasing 729 licenses. The ATA-286 works fine other > than this 1 issue. > > Do the Grandstream developers follow this list?? This problem has been > persistent for a LONG time and each new firmware version still has it > unfixed!! > > > On Sat, 14 Aug 2004 19:36:43 +0300, Lubomir Christov > <[EMAIL PROTECTED]> wrote: > > > > Yes, we are experiencing the same problem and because of that we > > switched the called HT ata to Cisco ATA 186 ... > > > > Lubo > > > > > > > > Andy Lee wrote: > > > [ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] > > > > > >>The problem I'm experiencing with many GS adapters, regardless of > > >>firmware version is this. Call from one phone to another phone using > > >>both the 'T' and 't' flags in the Dial() command. After they are > > >>connected, you should be able to press '#' on either phone to hear > > >>"transfer". What I am experiencing is the calling GS adapter will > > >>hear "transfer" when they press '#', but when the receiving GS adapter > > >>presses '#', nothing happens at all. Are you able to repeat this? If > > >>not, can you please tell me the firmware revisions and Asterisk > > >>version that you are using? > > >> > > >>Thank you very much. > > > > > > > > > Yes, I am experiencing the same problem. Works fine with the > > > BudgetTone phones but not with the HandyTone-286. > > > > > > Have you resolved this yet? We've been trying to figure this out on > > > and off for the past couple of weeks. > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > -- > > > > - > > Appradius Project: RADIUS authentication and accounting support for > > Asterisk PBX > > http://appradius.minitelecom.org/ > > - > > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sat, 14 Aug 2004, Andrew Kohlsmith wrote: > On Saturday 14 August 2004 18:29, Peter Svensson wrote: > > Using a BRI will eliminate echos from the pstn connection. Your ip phones > > should prevent echos from the local phone connections as well. That way > > you should not cause any noticable echo for the remote party. Being all > > digital has its advantages. :-) > > *wrong*, unless there is something drastically different between ISDN PRI and > ISDN BRI. > > I have *bad* echo on my Bell Canada PRI (via TE405P) -- software echo > cancellation helps but it's not perfect. All the BRI/PRI does is ensure that > *your end* is not generating echo, as there is no hybrid there. There is > still a hybrid in your handset, and there is still a hybrid at the other end > of the PSTN call. If you are _all digital_ as I wrote there is no hybrid in your handset. I agree that far-end echos can still occur but you are not casuing them. The remote party should not be hearing any echos. Canceling far-end echo is a hard thing to do properly. You can get nasty interactions between the echo canceler close to the echo source (where the canceling should be done) and your canceler. Also, you have no rough estimate of the echo characteristics to start off with so you have to train from scratch every time. (I think Asterisk always does this even for local lines where it probably could remember the line charateristics). The only good way to deal with far end echos are by keeping the latencies low enough. That way they sound like a pleasent reverb. For an all tdm voice path (we are lucky) this is no problem and no echo canceling is needed. For short hauls along non-congested links I think you can get low enough latencies for VoIP as well. We have made a few trials with VoIP and have had no annoying echos except when going across the atlantic with it and calling certain lines. Still, the other party heared no echos. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help - is voip good for in-house calls?
Network traffic really shouldn't be much of a concern if you have decent n-way switches installed -- pretty much all "mainstream" switches these days use n-way (Address Table, etc... Lots of names for the same technology). Those switches basically have enough smarts to know where the packets are going, and to isolate a datastream to the destination port. So if you have voice-traffic between ports 1 and 2, ports 3-16 won't even see it, so it doesn't bog down a whole network segment. In my setup (2xLinksys 4116 switches), I can stream video at full speed from my server to a ReplayTV unit without hearing so much as a click on an internal phone conversation -- the LEDs on the ports show traffic limited to the appropriate ports. QOS comes into play only when you have to route the voice-traffic over a WAN connection and it has to compete with data going over the same link. If you have a T1 coming into the office and place a VOIP call, then someone downloading a huge file needs to give up bandwidth to let the voice traffic through. > -Original Message- > From: Wiley E. Siler [mailto:[EMAIL PROTECTED] > Sent: Saturday, August 14, 2004 4:43 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Help - is voip good for in-house calls? > > Also, my HP switch has COS (class of service which is like > QOS) so I can prioritize the packets coming from my phones > over the standard network traffic. Even without this > switching feature turned on, performance was great. The > > -Original Message- > From: Francis Augusto Medeiros [mailto:[EMAIL PROTECTED] > Sent: Saturday, August 14, 2004 1:08 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Help - is voip good for in-house calls? > > My concerns are most latencies. Our network will be a switch > with lots of ports, all 100mb/s, with VERY low traffic. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List traffic/Software
Congratulations, you just volunteered. > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Saturday, August 14, 2004 2:04 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] List traffic/Software > > > I know this has been gone over before but > > It seems that most of the traffic to this list is the same 10 > questions > being asked over and over again. What if someone (with some > programing > skill) wrote a script so that when someone posted to the > list, it would > search the wiki and google and respond to them. If they > where not happy > with the response, then they could forward there message back > to the list > and it would go out to the 8000 people. While there are lots > of issues > with doing this, I do think it would cut down on the amount > of "newbie" > traffic that this list gets. > > Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help - is voip good for in-house calls?
On Sat, 14 Aug 2004, Jay Milk wrote: > QOS comes into play only when you have to route the voice-traffic over a > WAN connection and it has to compete with data going over the same link. > If you have a T1 coming into the office and place a VOIP call, then > someone downloading a huge file needs to give up bandwidth to let the > voice traffic through. The problems are mostly when you are on the road. All you need is one device with a queue of packets and you get a lot of latency. For a lan (or a wan under your control) latency should be manageable. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
