Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
At 11/3/03 6:57 PM, Anthony Wood <[EMAIL PROTECTED]> wrote: >Internals can use the IP address of the NAT box as the Asterisk Server >IP and then it should work. This doesn't work on my NAT box, unfortunately. Devices behind the NAT can't connect to the public IP address and talk to other devices behind the NAT. Don't know why (cheapo NAT box, most likely; it's part of my DSL modem), but I believe this situation is fairly common. -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod...
Hi Paul, - Original Message - From: "Paul Liew" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 12:39 AM Subject: Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod... > Hi Dan, > > Nice goingsome testing feedback. Testing your new client with > voicemail - after entering password (4 digits), 1 for new messages, then no > further digits can be sent. So far everything else OK. Someone else with this behaviour? I cannot reproduce it here. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Actiontec's Internet Phone Wizard and Digium's S100U
Hi, - Original Message - From: "Jim Flagg" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 1:28 AM Subject: Re: [Asterisk-Users] Actiontec's Internet Phone Wizard and Digium's S100U > > There is someone on this list who has the specification for the IPW driver? > > I want to provide full support for it in DIAX Phone, but Actiontec does not > > answer to my mails. > > > > Dan > > Dan, you might want to also look into the > Internet Power Phone 2000 - Standard Telephone Adapter > http://www.eutecticsinc.com/products/business.html > scroll to the bottom I know their products, but the problem is that now I only have the Actiontec's device. I am more interested to support S100U from Digium, if I can get a Windows driver for it. I'm interested to support too any USB davice available on the market who can provides the API calls required to access all the functionality. I don't want to have just the audio (as it is now with Actiontec's device). I want ring, on/off hook signaling, dialing, flash and so on. > > More than one person has sent e-mails to Actiontec asking about technical details > or drivers for the IPW but never got any response. I' m in the way to find myself how to use the drivers included in the PCPhone package from iConnect Here, but I still have a lon way to go. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 03, 2003 9:04 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) > > I was willing to give vb a chance at one time, but won't touch it any > > more. > > I am not defending VB. I won't touch it either (I use XWT for all my > cross-platform user interface type stuff, any web monkey can be proficient . I must admit that it was a lot more difficult to integrate the IAX library in a VB application than in a VC++ one (totally different way to pass things between them), but this is the nice part of this little project ;-) Best regards to you all, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer from Grandstream BT100?
Answer inline below.. Steven Sokol wrote: Hi, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 4a. Wait 5 seconds for the GS phone to actually dial the number you entered or hit the send/redial key. 4b. Call is transferred. 5. Grandstream hangs up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi Philipp,
Philipp von Klitzing a écrit:
Hi!
I have an asterisk box with one GS101 register to it in SIP mode and an
IP10S in MGCP mode.
I can dial IP10S from my GS101 and everything seems fine.
But from my IP10S I can't dial any number (GS or anything else).
Is that GS specific, or does the problem also include other SIP UAs like
X-Lite, X-Pro, SJPhone etc?
No it does not depend on the phone called. I am trying to make an IP to
PSTN gateway and I can't dial any number with my IP10S
Did you try canreinvite=no?
yes. Here is my mgcp.conf:
---
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 192.168.10.254
[192.168.10.10]
threewaycalling=yes
transfer = yes
callwaiting = no
callwaitingcallerid = yes
host = 192.168.10.10
nat = no
disallow = all
allow = g711
allow = alaw
callerid = toto <111>
line => aaln/1
-
You might try
Swissvoice support if the problem persists, they should have an intereset
to solve this ("rnc Info Lists" reported the same problem in a private
message earlier).
I will try it but I have understood that s/o has used IP10S with
success on this list so I have asked for it here before
In any case I'd be very interested to hear about your results, preferably
on this list. :-)
No problem
Regards,
Daniel
--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
RE: [Asterisk-Users] Actiontec's Internet Phone Wizard and Digium's S100U
Hi, > -Original Message- > There is someone on this list who has the specification for > the IPW driver? > I want to provide full support for it in DIAX Phone, but > Actiontec does not > answer to my mails. > > There is any windows driver available for the Digium's S100U > USB interface? > Anyone knows if it can provide full functionality for the > connected phone, > like callerid, callwaiting callerid, message indication, ring, etc.? I should think support for the S100U should be manageable and equally usefull ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod...
Hi, > -Original Message- > > Also, do you think its possible to block access to the > user/server-config > > part (maybe via a flag in the .ini file) ? It might confuse the less > > computer-able users :-) > I even think to a way to protect the config file to an > unauthorized external > editing. As you can see, the credentials are a little bit crypted (not > difficult to guess, but not for the normal user). > I can build a special release where you can edit credentials > only through a > separate tool. Actually, my wish would be to be able to create the package with correct (custom) credentials from the webserver, so that separate tool should be cross-platform (perl script ?). Being able to lock the ini file from editing is nice, but not a requirement for me. If people want to tinker around with it, that's fine with me. I only want to guard users who tinker around in the interface :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Daniel, the MGCP log you sent shows you sending the digits and asterisk receiving them, however after that either nothing happens (infinite digittimeout) or you cut the log short. Can you also send some console output with 'mgcp no debug' :-) It saves clutter. Maybe a peek at your extensions.conf might be usefull as well ? Also, can you tell us your phone's firmware ? (the IP10) I had one minor issue with the IP10 because of an older firmware version, a simple upgrade resolved it (by the way, in my case it was interpreting digits twice in some cases, i.e. dialling 326 would make asterisk think I was calling 33226) Best regards, Florian No it does not depend on the phone called. I am trying to make an IP to PSTN gateway and I can't dial any number with my IP10S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX softwarephone (for WIndows platform))
> On Mon, 2003-11-03 at 16:27, Alastair Maw wrote: >> On 03/11/03 20:03, Steven Critchfield wrote: >> >> > Sounds like you really need a C programmer and get into the guts >> > of asterisk. Can't get more flexible than having the source code >> > yourself to do anything you want. You could add your DSP routines into >> > the dsp.c file and call them when needed. You can also write a >> asterisk >> > application and have direct access to all the audio in every direction >> > just as you want it. >> >> But C isn't as maintainable as nice Java apps, and it's as simple as >> that. Basically, I'm after the most powerful interface possible to >> Asterisk, but trying to make it as friendly as possible to code things >> against. As far as our organization is concerned, that pretty much means >> Java objects. > > So you bought that line of Marketecture didn't you. I think there are > several large open source projects that prove that C is maintainable. > Maintainability is really a function of organization. If you can't be > organized, you will not produce very maintainable C code. > > I'll point out that I am not a C programmer, but making patches to > asterisk isn't that difficult. I have also made patches to the kernel > without too much hair pulling. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > Steve, You are right... Lots of proof that C is maintainable. I don't profess to be a C, VB or JAVA expert but have programmed for longer than I care to admit. What matters most is good solid and tight code regardless of the language. It all comes down to the number of CPU cycles needed to perform a given function. When doing real time processing, a few cycles here and a few there can add up to make a real difference. Object Oriented is nice for ease of writing/maintaining code but all of those objects have blocks of code behind them. A slight inefficiency there can really impact performance. Sure we have faster processors and lower cost memory every 6 months but thats no excuse for not writing the most efficient code possible. Asterisk does rather well on my Pentium 100/32 MB RAM. Wish I still had the Pentium 75 to try it on. It must really boogy on the bigger boxes. I contend that the "most powerful interface" is one that meets the requirements of the customer (1st requirement), is written to be the most efficient (2nd requirement) and maintainable (3rd requirement) as possible. The language to be used is the selection of the person doing the development. I'm not a fan of any Microsoft product but they do have a place in the world (for now). Kudos to Dan for his IAX phone. It works. He is responsive to bug fixes. Hopefully he will continue the development. Mark's offer of direct help I think speaks volumes about the importance of GPL IAX softphones for Win32. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, your MGCP log shows that asterisk receives the digits ok, but the log is cut short so I don't see asterisk dealing with the digits. Can you tell us more about your extensions.conf ? One final thing I can think of, what firmware version does your IP10 have ? I had one minor issue with older firmware, an upgrade resolved it easily. Florian No it does not depend on the phone called. I am trying to make an IP to PSTN gateway and I can't dial any number with my IP10S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Actiontec's Internet Phone Wizard and Digium's S100U
Hi, - Original Message - From: "Florian Overkamp" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 10:17 AM Subject: RE: [Asterisk-Users] Actiontec's Internet Phone Wizard and Digium's S100U >... > I should think support for the S100U should be manageable and equally > usefull ? You're right, but ufortunatelly I do not have it right now. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
> Daniel, > > the MGCP log you sent shows you sending the digits and asterisk receiving > them, however after that either nothing happens (infinite digittimeout) or > you cut the log short. Can you also send some console output with 'mgcp no > debug' :-) It saves clutter. Maybe a peek at your extensions.conf might be > usefull as well ? > > Also, can you tell us your phone's firmware ? (the IP10) > > I had one minor issue with the IP10 because of an older firmware version, > a > simple upgrade resolved it (by the way, in my case it was interpreting > digits twice in some cases, i.e. dialling 326 would make asterisk think I > was calling 33226) > > Best regards, > Florian > FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
On Mon, 3 Nov 2003 14:51:06 -0500 Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > > exten => _911,1,ChanIsAvail(Zap/1) > > exten => _911,2,Dial,Zap/1/911 > > exten => _911,3,Hangup() > > exten => _911,103,SoftHangup(Zap/1-1) > > exten => _911,104,Wait(1) > > exten => _911,105,Goto(1) > > That is pretty much exactly it, except I don't use _, and I play a file and > sleep(5) before what you're doing. > > > The only thing I would add is a variable to set for emergency call > > already on the channel. That way before you do the disconnect you can > > check to see if it is already an emergency call. > > I am not doing that currently but yes it would be a good thing, although > it's corner cases like that that I usually just document and leave as > is. :-) But isn't it likely that many people call 911 simultaneously in case of an emergency? Maybe it's not a corner case. Regards, Christian. > > Regards, > Andrew > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alert extensions without answering incoming call?
Hi, * gurus, I wonder if there is a way to alert the in-house extension(s) in case of an incoming external call without actually answering it, before somebody picks up the phone on one of the extensions? This way the caller wouldn't have to pay for the call until somebody answers. Regards, Christian Lademann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, > -Original Message- > FLorian, > What version of the IP10 firmware are you using?? I have > experienced the > multiple digit problem. Seems that this happens when dialing > more than 2 > digits. My 2 digit extensions seem to work fine but the ones > greater than > 2 digits get this repeating issue. I now have: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 217. 114. 96. 205 Mac address 00:05:90:02:03:0d Protocol MGCP 1.0 Best regards Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX softwarephone (for WIndows platform))
am i the only one getting tired of hearing about this? you are long time programmer, but not an expert of any of the listed languages below, yet you are 'professing' that they are languages that don't fit your 'requirements'. hmm oh and the 'quality assurance levels' (those p100/75 systems) are right in line with todays machines and requirements. there are alot of us that deal with lessor, but when you deal with a DSP device, you MAY want to use a machine with more horsepower. (just a suggestion) rnc Info Lists wrote: On Mon, 2003-11-03 at 16:27, Alastair Maw wrote: On 03/11/03 20:03, Steven Critchfield wrote: Sounds like you really need a C programmer and get into the guts of asterisk. Can't get more flexible than having the source code yourself to do anything you want. You could add your DSP routines into the dsp.c file and call them when needed. You can also write a asterisk application and have direct access to all the audio in every direction just as you want it. But C isn't as maintainable as nice Java apps, and it's as simple as that. Basically, I'm after the most powerful interface possible to Asterisk, but trying to make it as friendly as possible to code things against. As far as our organization is concerned, that pretty much means Java objects. So you bought that line of Marketecture didn't you. I think there are several large open source projects that prove that C is maintainable. Maintainability is really a function of organization. If you can't be organized, you will not produce very maintainable C code. I'll point out that I am not a C programmer, but making patches to asterisk isn't that difficult. I have also made patches to the kernel without too much hair pulling. -- Steven Critchfield <[EMAIL PROTECTED]> Steve, You are right... Lots of proof that C is maintainable. I don't profess to be a C, VB or JAVA expert but have programmed for longer than I care to admit. What matters most is good solid and tight code regardless of the language. It all comes down to the number of CPU cycles needed to perform a given function. When doing real time processing, a few cycles here and a few there can add up to make a real difference. Object Oriented is nice for ease of writing/maintaining code but all of those objects have blocks of code behind them. A slight inefficiency there can really impact performance. Sure we have faster processors and lower cost memory every 6 months but thats no excuse for not writing the most efficient code possible. Asterisk does rather well on my Pentium 100/32 MB RAM. Wish I still had the Pentium 75 to try it on. It must really boogy on the bigger boxes. I contend that the "most powerful interface" is one that meets the requirements of the customer (1st requirement), is written to be the most efficient (2nd requirement) and maintainable (3rd requirement) as possible. The language to be used is the selection of the person doing the development. I'm not a fan of any Microsoft product but they do have a place in the world (for now). Kudos to Dan for his IAX phone. It works. He is responsive to bug fixes. Hopefully he will continue the development. Mark's offer of direct help I think speaks volumes about the importance of GPL IAX softphones for Win32. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk does not hang up
hi, i am trying to do to autoattendant. here is my extension.conf part [tumpak] exten=>s,1,Dial,Zap/4|10 exten=>s,2,Voicemail,u exten=>s,102,Voicemail,b exten=>t,1,hangup so when a caller dials the extension 2 suppose, it enters to the above context.. everything is fine. the problem is when the caller hangs up the asterisk does not. after caller hangs up and tries again he always gets a busy tone until asterisk stops by timeout. i get this meesage in asterisk message is too long, ending it now... timeour zap/1-1 spawn extension(tumpak,t,1) exited non-zero on Zap1-1 earlier i tried it without the exten=>t,1,hangup line and i got no rule 't' in tumpak.. and added that . but i still get the same thing. can anyone help me with this. thanks cm = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone using * in a live production environment?
Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case study, just a few lines to try and get a grip on the number of real installations. Thank you :) Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash hook -> SIP device
Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I mean how does Asterisk see and handle this? What should the SIP device send to Asterisk so it works properly? regards Mickey Binder [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
Re: [Asterisk-Users] Red Alarm
On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz <[EMAIL PROTECTED]> wrote: > > > Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate > > > start > > > signaling), and just few seconds after this, all alarms are > cleared. > > > > > > This problem ocurrs many times/day, and if are calls in > > > progress, > > > these calls just hang-up. > > > Could it be an asterisk bug? Or may I contact the PSTN provider? > > > > I'd suggest your telco doing loopup and checking the circuit. > > My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Dan, problems discovered First, DIAX still crashes after hitting exit, and not only from system tray. Second, Why don't I get a busy signal? Even when I call myself. How many lines(calls) is DIAX capable of having concurrently? Third, I was playing with extensions.conf while adding users using DIAX, and suddenly, every call I made ringed back to me. I don't know if this is * fault, or DIAX fault. It's a great app ,Dan (the Man) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Sent: Tuesday, November 04, 2003 10:33 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) Hi, - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 03, 2003 9:04 PM Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform) > > I was willing to give vb a chance at one time, but won't touch it any > > more. > > I am not defending VB. I won't touch it either (I use XWT for all my > cross-platform user interface type stuff, any web monkey can be proficient . I must admit that it was a lot more difficult to integrate the IAX library in a VB application than in a VC++ one (totally different way to pass things between them), but this is the nice part of this little project ;-) Best regards to you all, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case study, just a few lines to try and get a grip on the number of real installations. Thank you :) Gavin. Hi Gavin, If its possible could I get a copy of the business case?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, Florian Overkamp a écrit: Hi, your MGCP log shows that asterisk receives the digits ok, but the log is cut short so I don't see asterisk dealing with the digits. Can you tell us more about your extensions.conf ? I have revisited my extensions.conf and seen that there were no context defined for my mgcp phones. So I have tried to define a proper context and define some dial plan for it in my extensions.conf. This didn't work. Next I have left the context blank in my mcgp.conf and modified the default dialplan in my extensions.conf and now my IP10S can dial out. Many thanks Florian for pointing this out. BTW is there some known issue with context keyword in chan_mgcp? One final thing I can think of, what firmware version does your IP10 have ? I had one minor issue with older firmware, an upgrade resolved it easily. Here is my info page: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 10. 10 Mac address 00:05:90:02:02:f0 Protocol MGCP 1.0 Daniel -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
[Asterisk-Users] Does anyone provide inbound UK numbers using IAX ?
Hi All, Is there anyone providing UK geographic numbers that can be terminated on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or 02, not 08xx). I've tried the sipcall.co.uk service and it looks very good when using X-Lite but it will not work with Asterisk. Switching to IAX should also resolve issues around NAT - hurray! -Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 10:04, WipeOut wrote: > Gavin Hamill wrote: > >I'm simply trying to add weight to the business case for new * installs, > >especially for those who have a very conservative management structure. > Hi Gavin, > > If its possible could I get a copy of the business case?? Ah, I can see how that sentence was miseleading... I'm talking rhetorically about the general "business case" for using * at all, as opposed to an actual document that I guess you thought I was referring to. If I already had something documented, it would certainly be publically available, since it benefits us all. Sorry :) gdh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone using * in a live production environment?
Me too! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, November 04, 2003 1:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Anyone using * in a live production environment? Gavin Hamill wrote: >Hullo again, all :) > >If you're using * to run telephony in a real business environment, can I >trouble you to write a short paragraph about the setup, and how you've >found the migration / daily use? > >I'm simply trying to add weight to the business case for new * installs, >especially for those who have a very conservative management structure. > >Like I say, I'm not looking for a case study, just a few lines to try >and get a grip on the number of real installations. > >Thank you :) > >Gavin. > > > Hi Gavin, If its possible could I get a copy of the business case?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
On Tue, 2003-11-04 at 10:14, Daniel ANDRE wrote: > Next I have left the context blank in my mcgp.conf and modified the > default dialplan in my extensions.conf and now my IP10S can dial out. Hullo Daniel :) Can I request that you post the pertinent parts of your config to the list, since I'm sure I'm not the only one who would benefit from a set of known-working configs for these phones. Personally, I'm on the verge of buying some SwissVoice handsets, simply because the mix of feature-set, price, and build quality seems to be untouchable. The GrandStreams are about the same price, but the build quality looks cheap and plastic - the IP10 actually looks like a business telephone. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Shoval, > Dan, problems discovered > > First, DIAX still crashes after hitting exit, and not only from system tray. All the time or after some specific operations? Did you registered with Asterisk server? > > Second, Why don't I get a busy signal? Even when I call myself. Ooops you're right... no busy signal...just hangup. I'll think about that ... > How many > lines(calls) is DIAX capable of having concurrently? 2 for the moment and you can switch between them using SELECT button (active only when you have 2 open calls) > > Third, I was playing with extensions.conf while adding users using DIAX, and > suddenly, every call I made ringed back to me. I don't know if this is * > fault, or DIAX fault. I think is * related (way you configure extensions.conf ..). > > It's a great app ,Dan (the Man) Thanks a lot, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone provide inbound UK numbers using IAX ?
We can. Feel free to contact me and let me know your requirement. Linus Magrathea - Original Message - From: "nathan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 10:18 AM Subject: [Asterisk-Users] Does anyone provide inbound UK numbers using IAX ? > Hi All, > > Is there anyone providing UK geographic numbers that can be terminated > on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or > 02, not 08xx). I've tried the sipcall.co.uk service and it looks very > good when using X-Lite but it will not work with Asterisk. Switching to > IAX should also resolve issues around NAT - hurray! > > -Nathan > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Gavin Hamill wrote: On Tue, 2003-11-04 at 10:04, WipeOut wrote: Gavin Hamill wrote: I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Hi Gavin, If its possible could I get a copy of the business case?? Ah, I can see how that sentence was miseleading... I'm talking rhetorically about the general "business case" for using * at all, as opposed to an actual document that I guess you thought I was referring to. If I already had something documented, it would certainly be publically available, since it benefits us all. Sorry :) gdh No problem, Maybe I should have read it closer.. Well then if you do put one together I would be interested in seeing that one.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem in running .gsm file
Hi I am already succesfully installed asterisk in my Linux Box. And successful to transfer call ATA-186>>Asterisk>>Mediatrix 1204 switch>>PSTN with the default IVR features of Asterisk. But I can't hear the .gsm file.I already istalled "zgsmplay-1.3". Any body please help me to play these .gsm file. Thanks in advance. Dipak Kumar Paul _ Discover digital jadoo! Enter this contest. http://server1.msn.co.in/sp03/mediacenter/index.asp Win cool prizes! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Florian Overkamp wrote: Hi, -Original Message- FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. I now have: Phone name Undefined why u did not define the name for this phone ? - it seams that name will be used as gw name in mgcp ... I'm about @[ip] Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 217. 114. 96. 205 Mac address 00:05:90:02:03:0d Protocol MGCP 1.0 Best regards Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer from Grandstream BT100?
On Mon, 3 Nov 2003, John Brown (CV) wrote: > what version of GS firmware are you running ? > > > I call from PSTN to GS, GS does xfer to XTEN, hang up GS > call continues > > if you aren't running 1.0.3.81 or newer, then upgrade :) Or NEWER Latest I can find is 1.0.3.81. Care to give us all an early holiday gift? :) > > john brown > chagres > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compil error with Mandrake pwlib
hi , Does anyone work with Mandrake 9.0 and have try to install pwlib ? I have some compil errors with statsfs.h : In file included from /usr/include/sys/statfs.h:26, from /usr/include/sys/vfs.h:4, from osutil.cxx:217:/usr/include/bits/statfs.h:26: redefinition of `struct statfs'/usr/include/asm/statfs.h:12: previous definition of `struct statfs'make[3]: *** [/root/pwlib/lib/obj_linux_x86_d/osutil.o] Erreur 1make[3]: Quitte le répertoire `/root/pwlib/src/ptlib/unix'make[2]: *** [debug] Erreur 2make[2]: Quitte le répertoire `/root/pwlib'make[1]: *** [libs] Erreur 2make[1]: Quitte le répertoire `/root/pwlib'make: *** [debuglibs] Erreur 2 Regards Rattana BIV Rattana3 place Charles Hernu69100 Villeurbanne[EMAIL PROTECTED]Tel 04 72 83 76 80Fax 04 72 83 76 89 small_logo.gif Description: Binary data
[Asterisk-Users] ipphone voicemail problems
Im having a little problem with voicemail and my cisco phones i was wondering if anyone might have seen this before and let me know whats going on. it spits this out and then my cisco ip phone reboots im using the latest cvs and a cisco 7910 phone WARNING[1234379840]: File res_adsi.c, Line 205 (__adsi_transmit_messages): Unable to send CAS -- Playing 'vm-login' (language 'en') Skinny [EMAIL PROTECTED] went on hook WARNING[1234379840]: File app_voicemail.c, Line 1907 (vm_execmain): Couldn't read username == Spawn extension (internal, 850, 1) exited non-zero on 'Skinny/[EMAIL PROTECTED]' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk does not hang up
Hi, Try: - exten=>t,103,hangup or exten=>s,103,hangup Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of C M Sent: 04 November 2003 09:37 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk does not hang up hi, i am trying to do to autoattendant. here is my extension.conf part [tumpak] exten=>s,1,Dial,Zap/4|10 exten=>s,2,Voicemail,u exten=>s,102,Voicemail,b exten=>t,1,hangup so when a caller dials the extension 2 suppose, it enters to the above context.. everything is fine. the problem is when the caller hangs up the asterisk does not. after caller hangs up and tries again he always gets a busy tone until asterisk stops by timeout. i get this meesage in asterisk message is too long, ending it now... timeour zap/1-1 spawn extension(tumpak,t,1) exited non-zero on Zap1-1 earlier i tried it without the exten=>t,1,hangup line and i got no rule 't' in tumpak.. and added that . but i still get the same thing. can anyone help me with this. thanks cm = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, You seem to have nice recent firmware. I am not aware of any issues with the context configuration with MGCP, it all seems to work just fine for me. Strange... Best regards, Florian I have revisited my extensions.conf and seen that there were no context defined for my mgcp phones. So I have tried to define a proper context and define some dial plan for it in my extensions.conf. This didn't work. Next I have left the context blank in my mcgp.conf and modified the default dialplan in my extensions.conf and now my IP10S can dial out. Many thanks Florian for pointing this out. BTW is there some known issue with context keyword in chan_mgcp? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX2 Java library
On 04/11/03 04:59, Steven Critchfield wrote: But C isn't as maintainable as nice Java apps, and it's as simple as that. Basically, I'm after the most powerful interface possible to Asterisk, but trying to make it as friendly as possible to code things against. As far as our organization is concerned, that pretty much means Java objects. So you bought that line of Marketecture didn't you. I think there are several large open source projects that prove that C is maintainable. Maintainability is really a function of organization. If you can't be organized, you will not produce very maintainable C code. Stop it right there. This mailing list isn't the place for language holy wars. Java is the tool of choice for our organization, and therefore Java code is more maintainable within it. Everyone here talks Java. Not everyone talks C. If we hire less good programmers, Java hold their hand and enforces structure in a way that C doesn't. I'm not arguing that decent C code isn't maintainable. I'm stating that in our particular case, Java fits the bill better. It's that simple. As for this: It all comes down to the number of CPU cycles needed to perform a given function. When doing real time processing, a few cycles here and a few there can add up to make a real difference. Object Oriented is nice for ease of writing/maintaining code but all of those objects have blocks of code behind them. A slight inefficiency there can really impact performance. Sure we have faster processors and lower cost memory every 6 months but thats no excuse for not writing the most efficient code possible. You're entirely wrong. Provided it's fast enough, it's all about maintainability, extensibility and scalability. The bottom line is that we're in it to make money. If it runs on hardware that's $100 cheaper, but takes four times as long to develop, then total cost is much higher. If it's easy to repurpose it at a later date, or easier to have someone unfamiliar with the codebase to come along and customize it, that's worth a lot. Both are significant, but maintainability is much more important than performance, as that's where the real cost lies. -- Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, > -Original Message- > > > >Phone name Undefined > > > > > why u did not define the name for this phone ? - it seams > that name will > be used as gw name in mgcp ... I'm about @[ip] > Interesting. Actually I never defined it because it was not needed in my setup. Asterisk and the phone understand eachother just fine like this. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] high system load running asterisk
hi! i did install asterisk on a freebsd box. after removing the codec_mp3_d.so module it starts without SIGSEGV. -> i didn't care about this issue. however while running asterisk without any clients online my system-load is about 1 -> and asterisks seems to run all the time. is this the default behavior? is there any interface or command line command to find out which of the modules takes all the cpu-time? best regards christoph loibl -- CHRISTOPH LOIBL > mailto:[EMAIL PROTECTED] | A red sign on the door of a physics professor: http://pix.tix.at | "If this sign is blue, you're going too fast." CHL-RIPE > PGP-Key-ID: 0x4B2C0055 >>> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX2 Java library
On 03/11/03 23:31, Jeremy McNamara wrote: - The documentation for AGI is very poor. I know it is for IAX, too, but I can see a Java IAX library being useful for client development too, and I'd like to give a little back to the * community, you know? You have the source code. What more do you need? Seeing as you're asking, an RFC would be nice. But then, this has already been discussed. Maybe you had your blinkers on so tight you didn't notice. :) For those who are interested in such things, if I were to write an RFC proposal, would anyone actually take the time to check it and do some proof-reading (who is qualified to do this for IAX2? Is it only really Mark? Or are there others out there who are experts?). If people are interested in this, I'll try to document things as I go along... Regards, -- Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high system load running asterisk
On 04/11/03 11:33, Christoph Loibl wrote: however while running asterisk without any clients online my system-load is about 1 -> and asterisks seems to run all the time. is this the default behavior? No. Load average on an idle Asterisk box should be very close to zero. I don't think there's an easy way to find which module is using all the CPU time. You can add a noload command to the modules.conf file for each module in turn until you've pinned it down. Regards, -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rollout tips
Hi" > Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and > many other places (like our branch office, my home dial-up account, my > parents dial-up account) It's been fine for me (Gin ermany, both via university (GWin) and T- Online), no problems at all, so it appears this is a "regional" DSN problem in your area? Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
This looks to me like the approach that Pingtel took for NAT. I think it is a good option to have but having STUN as an additional option is really what we want. You can find an implementation of a STUN library and apps at www.vovida.org. The External IP approach has some flaws and can be a pain to configure for people that do not know what is actually being done with this data. I will try to explain this since I have to test this stuff on vendor phones every day... SIP is a text-based protocol which means that address information is embedded in each SIP message as "text". Unfortunately, most routers, etc. do not have a SIP ALG so the address information in the UDP or TCP connections get corrected through the NAT function, the payload which in this case is SIP and SDP (RTP setup messages) do not get translated. The other end of the call outside your private network sees your private IP addresses and cannot route to them. External IP basically says "put this address in the SIP and SDP messages instead of my private address". The problem here is that if your lease is up on your ISP connection and the renew gives you another address, you're out of business until you update your settings. The other thing is, you must port forward your SIP port (usually 5060) and every incoming RTP/RTCP port pairs from the NAT router to Asterisk. STUN is pretty simple and works well. This feature actually queries a STUN server on the public side and askes what does your external IP and port look like. It also determines the level of IP security that your are using. (Read the RFC on STUN, it is usefull) You don't have to port forward anything because STUN enabled devices take advantage of the ALG in most firewalls that maps incoming traffic back to the app (Asterisk in this case) if the packets arrive at the same address/port that packets just went out. If the connection is idle for more than a set number of seconds, the mapping is automatically deleted. This is why you see the devices "pinging" each other every so often. This allows an incoming call to reach the SIP port. Having BOTH External IP and STUN would give us the greatest flexibility because if we didn't have a STUN server on the other end we could manually set it. Martin Pycko wrote: It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Martin On Mon, 3 Nov 2003, WipeOut wrote: Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is "externip" and new parameter?? Does it do a similar thing for the server as what "nat=yes" does for the phone? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
> First, DIAX still crashes after hitting exit, and not only from system tray. All the time or after some specific operations? Did you registered with Asterisk server? [Shoval Tomer] Every time I exit, and I have registered successfully. > > Second, Why don't I get a busy signal? Even when I call myself. Ooops you're right... no busy signal...just hangup. I'll think about that ... [Shoval Tomer] it doesn't hangup, it actually rings, as if I could answer myself... I'll retest and get back to you. > How many > lines(calls) is DIAX capable of having concurrently? 2 for the moment and you can switch between them using SELECT button (active only when you have 2 open calls) [Shoval Tomer] Does this mean you will only get a busy signal if a third call comes in? > > Third, I was playing with extensions.conf while adding users using DIAX, and > suddenly, every call I made ringed back to me. I don't know if this is * > fault, or DIAX fault. I think is * related (way you configure extensions.conf ..). [Shoval Tomer] I think you are correct. [Shoval Tomer] Thanks. > > It's a great app ,Dan (the Man) Thanks a lot, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Knowledge Sharing on Asterisks issue.
Hello All I was able to successfully transfer calls in the following Phases. ATA-186>>Asterisk>>Mediatrix 1204 switch>>PSTN ATA-186>>Asterisk>>VOCAL server>>Mediatrix 1204 switch>>PSTN PSTN>>Mediatrix 1204 switch>>Asterisk>>VOCAL Server>>Mediatrix 1204 switch>>PSTN Now I have to implement more features in this setup 1. IVR authentication. 2. User ID. 3. User account. 4. User money account. 5. User Database. Please help me to setup these features. Thanks in advance. Dipak Kumar Paul. _ Make fun prints Click here to order. http://www.kodakexpress.co.in Delivery anywhere in India. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high system load running asterisk
hi! On Tue, Nov 04, 2003 at 11:46:40AM +, Alastair Maw wrote: > On 04/11/03 11:33, Christoph Loibl wrote: > >however while running asterisk without any clients online my system-load is > >about 1 -> and asterisks seems to run all the time. is this the default > >behavior? > > No. Load average on an idle Asterisk box should be very close to zero. > I don't think there's an easy way to find which module is using all the > CPU time. You can add a noload command to the modules.conf file for each > module in turn until you've pinned it down. thats what i did. thanks for your help. the module in question was: pbx_wilcalu.so i don't really know what it is needed for (since i'm in a very early stage of evaluating *) however my sip-voip-phone still has access to the demo-application. thanks for your help. regards, christoph loibl -- CHRISTOPH LOIBL > mailto:[EMAIL PROTECTED] | A red sign on the door of a physics professor: http://pix.tix.at | "If this sign is blue, you're going too fast." CHL-RIPE > PGP-Key-ID: 0x4B2C0055 >>> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi, > > Second, Why don't I get a busy signal? Even when I call myself. > Ooops you're right... no busy signal...just hangup. > I'll think about that ... > [Shoval Tomer] it doesn't hangup, it actually rings, as if I could answer > myself... I'll retest and get back to you. There is no busy signal in the actual version. > > > How many > > lines(calls) is DIAX capable of having concurrently? > 2 for the moment and you can switch between them using SELECT button (active > only when you have 2 open calls) > [Shoval Tomer] Does this mean you will only get a busy signal if a third > call comes in? I'll test that. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:asterisk does not hang up
Dave i tried that. it didn't work. = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi! > BTW is there some known issue with context keyword in chan_mgcp? For the sake of documentation: chan_mgcp doesn't reload configs on 'reload' http://bugs.digium.com/bug_view_page.php?bug_id=268 Since that bug has been resting there for some time now it might be good if anyone else that has made the same experience would add a comment to that bug to increase its weight...?! Secondly a question: From the two sources I come to understand that is *is* possible to run an MGCP phone behind NAT (opposed to what Florian stated earlier on this list)? My order of an ip10s is going out today, but maybe some of you MGCP folks can give this a try already now and report back? Finally I think someone should open a tiny bug note for a better sample mgcp.conf that comes with * - what do you think? Thanks, Philipp http://bugs.digium.com/bug_view_page.php?bug_id=129 add the option to prevent native bridge - canreinvite sometime I need to prevent to create native bridge in chan_mgcp [Quote from an archived message on this list:] After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This works: nat=yes line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 This doesn't: line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 nat=yes This makes sense if lines were treated as individual channels through NAT, but they aren't. NAT capability is dictated by the Gateway itself, and not each endpoint/subchannel. I hope this saves somebody some time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high system load running asterisk on FreeBSD
Christoph Loibl wrote: however while running asterisk without any clients online my system-load is about 1 -> and asterisks seems to run all the time. is this the default behavior? Check the list archives at http://lists.digium.com I reported the same problem a while ago and got no solution, but a lot of helpful advice from friends on the list on how to track it down. Someone suggested a problem with pthreads and that's likely. However, I could not manage solving it myself. Any ideas or help solving this is appreciated! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rollout tips
Philipp von Klitzing wrote: Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many other places (like our branch office, my home dial-up account, my parents dial-up account) It's been fine for me (Gin ermany, both via university (GWin) and T- Online), no problems at all, so it appears this is a "regional" DSN problem in your area? Jim recently added another DNS, so we hope that even non-europeans :-) will have better access now. If not, keep us posted. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, > -Original Message- > Secondly a question: From the two sources I come to > understand that is > *is* possible to run an MGCP phone behind NAT (opposed to > what Florian > stated earlier on this list)? My order of an ip10s is going > out today, > but maybe some of you MGCP folks can give this a try already now and > report back? Hmm, now that would be very welcome indeed Someone please prove me wrong on this account :-)) > Finally I think someone should open a tiny bug note for a > better sample > mgcp.conf that comes with * - what do you think? Feel free to build one :-) > [Quote from an archived message on this list:] > After spending some time trying to get a DG-104S working behind NAT, > I finally found the problem. > > I made the incorrect assumption that nat=yes in mgcp.conf works just > like sip.conf. The channels within a gateway are treated more closely > to zap channels than sip channels (from a .conf standpoint). > > What this means is that you have to put nat=yes BEFORE any > subchannel definitions: > > This works: > > nat=yes > line => aaln/1 > line => aaln/2 > line => aaln/3 > line => aaln/4 > > This doesn't: > > line => aaln/1 > line => aaln/2 > line => aaln/3 > line => aaln/4 > nat=yes > > This makes sense if lines were treated as individual channels through > NAT, but they aren't. NAT capability is dictated by the > Gateway itself, and > not each endpoint/subchannel. Hmmfun. I may try this, but not before the end of the week... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *, Fritz!PCI and strange behavior
I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI (chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone else has seen them: - Now and then, * just exits. Until now I had lowish-level verbosity on, so all I saw was 'Executing last minute cleanups'. What can trigger * exits? (in other words, what should I pay attention to when attempting to debug this?) - Very often, after * runs for a while, it stops recognizing incoming ISDN calls and refuses to send out ISDN calls. The funny thing is, restarting * or CAPI doesn't work - I have to shutdown both, unplug and replug the ISDN cable, and then after startup everything works again. At first, I thought that it might be a bad cable, so I taped down everything in order to prevent it from moving. This didn't help. I really do not understand why the thing with the cable is necessary. Any light that you can shine on this would be most helpful. OBTW: the answer "don't use a Fritz" is not applicable here - I'm trying to assess the feasibility of making a <300$ ISDN SoHo PBX... -- Cees de Groot http://www.tric.nl <[EMAIL PROTECTED]> tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Red Alarm
How far is your server from the telco box? I found that with extended distances, my reliabilty was significantly decreased. If you still have problems, check your RJ-48X jack for connection problems. -sb -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 5:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Red Alarm On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz <[EMAIL PROTECTED]> wrote: > > > Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate > > > start > > > signaling), and just few seconds after this, all alarms are > cleared. > > > > > > This problem ocurrs many times/day, and if are calls in > > > progress, > > > these calls just hang-up. > > > Could it be an asterisk bug? Or may I contact the PSTN provider? > > > > I'd suggest your telco doing loopup and checking the circuit. > > My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] *, Fritz!PCI and strange behavior
Hi Cees, I have a similar setup, I don't use zaprtc on that machine though, it also has a X100P to deal with timings. > -Original Message- > - Now and then, * just exits. Until now I had lowish-level > verbosity on, > so all I saw was 'Executing last minute cleanups'. What can trigger > * exits? (in other words, what should I pay attention to when > attempting to debug this?) This I have not seen myself. > - Very often, after * runs for a while, it stops recognizing incoming > ISDN calls and refuses to send out ISDN calls. The funny thing is, > restarting * or CAPI doesn't work - I have to shutdown both, unplug > and replug the ISDN cable, and then after startup everything works > again. At first, I thought that it might be a bad cable, so I taped > down everything in order to prevent it from moving. This > didn't help. > I really do not understand why the thing with the cable is > necessary. I have seen this too. Freakishly enough rebooting the machine is not good enough, but leaving it powered down for a night works good too - it saves the cable trick ;-) Still, I too would like to know what causes this, but I have no idea where to start on it... > Any light that you can shine on this would be most helpful. OBTW: the > answer "don't use a Fritz" is not applicable here - I'm > trying to assess > the feasibility of making a <300$ ISDN SoHo PBX... Wait another while for KPJ to finish the ZapBRI stuff ? :-) By the way, X100P's work like a charm for this, except for dutch callerid... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Actiontec's Internet Phone Wizard and Digium's S100U
> I am more interested to support S100U from Digium, if I can get a Windows > driver for it. If you are serious about doing this, and you are willing to LGPL or GPL your final product (remember, even GPL'ing your code, you can still licenses it commercial as we are able to do with Asterisk), I can make this happen. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hello, I have experienced the 2 digits problem earlier. Here is my "old" configuration: Phone name Undefined Appli version IP10 M v0.2.0 (Build1) Boot version IP10 Boot v0.2.0 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 10. 11 Mac address 00:05:90:02:02:38 Protocol MGCP 1.0 With the new software version this pb disappeared: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 10. 10 Mac address 00:05:90:02:02:f0 Protocol MGCP 1.0 Regards, Daniel Florian Overkamp a écrit: Hi, -Original Message- FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. I now have: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 217. 114. 96. 205 Mac address 00:05:90:02:03:0d Protocol MGCP 1.0 Best regards Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec?
> > >> > > > > > >Speex works perfect with IAX but not that crack headed > x-lite stuff. > > > Can anyone make any recommendations, from personal experience, on a good softphone that has good look and feel, and of course reasonable sound quality, and works with Asterisk? Ray [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel Performance Primitives
Hi Ernest, I tried IPP, but couldn't get much performance out of it. When I tried diassembling one or two routines to see what they looked like, there seemed at be a llo of overhead in the routines that destroyed all the benefits. Regards, Steve Ernest W. Lessenger wrote: Hey all, For those of you who are really worried about asterisk performance, I thought I might alert you to a "toy" you might play around with. The Intel Performance Primitives contain a number of optimized functions for use in digital signal processing that could help with echo cancellation, codec transformations, etc. I don't have any idea how useful this would be in Real Life (actual performance gain, license compatibility, etc), but there you go... http://www.intel.com/software/products/ipp/ipp30/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Andrew Gillham wrote: Steve Underwood wrote: Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve I was measuring about 32-36 Kbit/s for a G.729 call, and around 20-22Kbit/s for G.723. This is at the DSL router, so it includes all of the overhead. If you're on a dialup modem, that can make or break the call. -Andrew What you are seeing their is not the codec at work, but the huge overhead of RTP. G.723.1 uses 30ms blocks. G.729 normally uses 20ms blocks. They are both usually sent with one block in each RTP packet. If you don't mind a little more latency, put two G.729 blocks into each RTP packet (I'm not sure if Asterisk supports that, but many things do). Then you have a packet every 40ms, a lower bit rate than G.723.1 and better voice quality. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4 X100P's, 4 7960's Same Box?
Cisco's CM can do the multiple line appearances, showing state. We could not ever get ringback on any phone except the extension that it was assigned to. Our workaround was to get a sidecar for each handset. With the side cars, you get colored lights that blink different colors for different call states. When it rang back, it would blink orange (I think) which was a visual cue that it needed to be picked up. I have not yet tried this with *, as I just left BYU to go to a small company with an annual IT budget of about a burger and 2 fries. But I know the skinny protocol is going to support it. There may be some features we need to fill out in the code for * to be able to take advantage of those features. Hope this helps, David Gomillion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, November 03, 2003 10:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 4 X100P's, 4 7960's Same Box? > I'm going to use Cisco 7960's for the phones; is there a better phone I > should be using? excellent choice. > I need to know if this is possible... > > On each phone program the appearances of 4 "Extensions" that are really > the 4 phone lines? yes for inbound. 1 rings 1... 2 rings 2.. and so on. Not the best method of doing this.. but ok. > If Phone "A" puts line 1 on hold can phone "B" see it on hold or will it > just show off hook? Not possible with SIP... might be with Skinny. Use parking or transfer. > Can the phones ring back (Even the one that did not put the call on > hold) if it's not picked up in 30 or so seconds? Nope. (See previous answer) > If phone A is using line 1 and phone B is using line 2 will phone C see > the other 2 call appearances? Nope. (Not with SIP) > MOST IMPORTANT can I use 4 X100P's in the same box? What can I expect? > >From what I read in the posts the X100P is a $10 modem marked up to pay > for the Asterisk dev, and I don't mind paying for that I just want to > make sure they are reliable and will get the job done, they will get a > good workout for about 3 hours a day! I have 3 in my Dell Poweredge 500SC but you need to have a good bios. One that will let you put each card on its own IRQ. > This email transmission and any attachments are for the sole use of the > intended recipient(s) and may contain confidential and privileged > information that is the sole property of PathAxis, Inc. Any unauthorized > review, use, disclosure or distribution is prohibited. If you are not > the intended recipient, please contact the sender and destroy and delete > all copies of this email and any attachments. These warnings are stupid. They should be at the top if you really want people to read them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
An E1 can be a long way from the box with the right cable. However many people use the wrong cable. Using a LAN cable for an E1 often gives errors if the cable is more than just a few metres long. Although the plugs look the same, the twisted pairs should be grouped differently in an E1 cable, and it really makes a difference. If the drop cable is only a couple of metres long, a LAN cable is usually adequate. This is also true for T1s. Regards, Steve Bisker, Scott (7805) wrote: How far is your server from the telco box? I found that with extended distances, my reliabilty was significantly decreased. If you still have problems, check your RJ-48X jack for connection problems. -sb -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 5:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Red Alarm On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz <[EMAIL PROTECTED]> wrote: Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? I'd suggest your telco doing loopup and checking the circuit. My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec?
Hi Ray - Original Message - From: "Ray Burkholder" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 4:04 PM Subject: RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec? > Can anyone make any recommendations, from personal experience, on a good > softphone that has good look and feel, and of course reasonable sound > quality, and works with Asterisk? > > Ray Have you tried DIAX? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Tue, 04 Nov 2003 22:14:17 +0800 Steve Underwood <[EMAIL PROTECTED]> wrote: > An E1 can be a long way from the box with the right cable. However > many people use the wrong cable. Using a LAN cable for an E1 often > gives errors if the cable is more than just a few metres long. > Although the plugs look the same, the twisted pairs should be grouped > differently in an E1 cable, and it really makes a difference. If the > drop cable is only a couple of metres long, a LAN cable is usually > adequate. This is also true for T1s. I changed the LAN cable (about 5 meters). Now, asterisk is connected with a 1.5m cable. I hope this help. Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: *, Fritz!PCI and strange behavior
Florian Overkamp <[EMAIL PROTECTED]> said: >> Any light that you can shine on this would be most helpful. OBTW: the >> answer "don't use a Fritz" is not applicable here - I'm >> trying to assess >> the feasibility of making a <300$ ISDN SoHo PBX... > >Wait another while for KPJ to finish the ZapBRI stuff ? :-) By the way, >X100P's work like a charm for this, except for dutch callerid... > X100P? I'm talking about a PBX that does BRI at the outside and (probably) SIP phones on the inside - what role would an X100P play in such a setup? (kapejod - I forgot what hardware ZapBRI is target at; similar price level as Fritz!?) Anyway, glad that I'm not the only one that has to do the cable thingy - at the very least it serves as a confirmation that I'm not as crazy as I like to think now and then :-) -- Cees de Groot http://www.tric.nl <[EMAIL PROTECTED]> tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
-- Original Message -- From: Gavin Hamill <[EMAIL PROTECTED]> >Hullo again, all :) > >If you're using * to run telephony in a real business environment, can I >trouble you to write a short paragraph about the setup, and how you've >found the migration / daily use? > >Thank you :) > >Gavin. > Ok here is a short paragraph on our use of Asterisk in the real world. 1 inbound PRI ISDN 23/24 channel - Local phone service with 60 DID numbers. 1 Long Distance T1 line for inbound 800 numbers and all outbound long distance calls. running on P4 1.7G 512mg 20 gig HDD with 2 T400P boards. Ethernet on MB Intel 845G board. Hardware: 4 Adtran 750 with 24 FXS channels each. 1 Adtran 600 with 4 FX0 and 12 FXS ports. 1 ZetaFax server with US robotics modem. 1 HP Fax as backup 4 Inbound RAS lines for users 2 outbound RAS modems for dial out support lines. 40 452 phones (Really bad choice for phones) 10 390 phones (Again better then 452 but still bad phones) Cisco ATA 186 (nice works great) Cisco 7960 (Nice phone but worst phone to setup and maintain) 4 SIP phones X-Ten Lite with Telex USB connection to PC (Works great) Overall system is working with Support queues(AGI login user accounts) and meeting rooms. Voicemail system is not very good need some way to configure the boxes. They really need to redo this application for more standard settings. We have MOH working without any problems. Major down is no Graphical interface. No actual working manager. Got to get them to fix the Zombie lines. (I feel it's mainly do to our 390 and 452 phones. It works needs some fine tuning but it works. I have nothing good to say about the Aastra phones 390 or the 452. They are not really good for heavy use like we need! The Cisco 7960 is nice to look at but in the real world it's hard to get working and setup. If you don't know about Linux or are able to use scripts it's a real mess to keep up! This is where it's being held back as a real world player! This is the basic setup. Next step is outside offices connection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: *, Fritz!PCI and strange behavior
Hi, > -Original Message- > >Wait another while for KPJ to finish the ZapBRI stuff ? :-) > By the way, > >X100P's work like a charm for this, except for dutch callerid... > > > X100P? I'm talking about a PBX that does BRI at the outside and > (probably) SIP phones on the inside - what role would an X100P play > in such a setup? You could use one or two X100P's to get lines in, instead of ISDN. > (kapejod - I forgot what hardware ZapBRI is target at; > similar price level > as Fritz!?) For now, custom cards (quadBRI), eventually: HFC PCI cards (cheap stuff) > Anyway, glad that I'm not the only one that has to do the > cable thingy - > at the very least it serves as a confirmation that I'm not as crazy as > I like to think now and then :-) H, that's still debatable :-) Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: *, Fritz!PCI and strange behavior
> I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI > (chan_capi 0.3.0), and have a couple of funny things - I wonder if > anyone else has seen them: > > - Now and then, * just exits. Until now I had lowish-level > verbosity on, > so all I saw was 'Executing last minute cleanups'. What can trigger > * exits? (in other words, what should I pay attention to when > attempting to debug this?) I have seen Asterisk spontaneously exit very occasionally, but not at this particular point. > - Very often, after * runs for a while, it stops recognizing incoming > ISDN calls and refuses to send out ISDN calls. I have this. If I try to dial out, I get an "all channels are busy at this time" error, when they are not. >The funny thing is, > restarting * or CAPI doesn't work - I have to shutdown both, unplug > and replug the ISDN cable, and then after startup everything works > again. At first, I thought that it might be a bad cable, so I taped > down everything in order to prevent it from moving. This > didn't help. > I really do not understand why the thing with the cable is > necessary. Straight asterisk restart always clears this condition for me. > Any light that you can shine on this would be most helpful. OBTW: the > answer "don't use a Fritz" is not applicable here - I'm trying to > assess the feasibility of making a <300$ ISDN SoHo PBX... I think the problem may be related to call progress indication from the ISDN line. I have UK ISDN2e (packaged as Business Highway - which includes what is effectively a telco-owned TA with two analogue ports). I have noticed that outgoing channels getting tied up corresponds to placing a call which is terminated prematurely (e.g. hangup before completing dialing) or dialing a call which can't be completed because the dialed PSTN subscriber number is invalid. I've learned to live with it - but it would be great to get to the bottom of it. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP10S and Handset
With some lively IP10S discussions here maybe someone knows about this issue: I can use the speaker phone ok. However the handset and switch hook do not seem to work. If I enable "headset" then I can get audio via the handset but still have to use the speaker phone button to take ot "off hook". Seems a bit wierd.. I have sent it to Swissvoice but no answer back yet. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 03:48, Gavin Hamill wrote: > Hullo again, all :) > > If you're using * to run telephony in a real business environment, can I > trouble you to write a short paragraph about the setup, and how you've > found the migration / daily use? > > I'm simply trying to add weight to the business case for new * installs, > especially for those who have a very conservative management structure. > > Like I say, I'm not looking for a case study, just a few lines to try > and get a grip on the number of real installations. I'm not trying to flame you for this message, but this is something that is asked about quite often and doesn't exactly prove anything. Just because I successfully pulled off an installation doesn't mean you will be successful. There is obviously working systems out here otherwise there wouldn't be this much traffic on this list. Maybe what needs to happen to keep this question from coming up over and over again would be to get this on the wiki. Someone want to create the page and then post the link to that specific page to be filled in? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high system load running asterisk
On Tue, 2003-11-04 at 06:41, Christoph Loibl wrote: > hi! > > On Tue, Nov 04, 2003 at 11:46:40AM +, Alastair Maw wrote: > > On 04/11/03 11:33, Christoph Loibl wrote: > > >however while running asterisk without any clients online my system-load is > > >about 1 -> and asterisks seems to run all the time. is this the default > > >behavior? > > > > No. Load average on an idle Asterisk box should be very close to zero. > > I don't think there's an easy way to find which module is using all the > > CPU time. You can add a noload command to the modules.conf file for each > > module in turn until you've pinned it down. > > thats what i did. thanks for your help. the module in question was: > > pbx_wilcalu.so > > i don't really know what it is needed for (since i'm in a very early stage > of evaluating *) however my sip-voip-phone still has access to the > demo-application. ahhh, This is the module that picks up the sample.call files and makes phone calls for you. >From reading over some of the patches for BSD lately, I think there is a problem with the way polling is done on the directory. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Ariel Batista wrote: Ok here is a short paragraph on our use of Asterisk in the real world. 1 inbound PRI ISDN 23/24 channel - Local phone service with 60 DID numbers. 1 Long Distance T1 line for inbound 800 numbers and all outbound long distance calls. running on P4 1.7G 512mg 20 gig HDD with 2 T400P boards. Ethernet on MB Intel 845G board. Hardware: 4 Adtran 750 with 24 FXS channels each. 1 Adtran 600 with 4 FX0 and 12 FXS ports. 1 ZetaFax server with US robotics modem. 1 HP Fax as backup 4 Inbound RAS lines for users 2 outbound RAS modems for dial out support lines. 40 452 phones (Really bad choice for phones) 10 390 phones (Again better then 452 but still bad phones) Cisco ATA 186 (nice works great) Cisco 7960 (Nice phone but worst phone to setup and maintain) 4 SIP phones X-Ten Lite with Telex USB connection to PC (Works great) Overall system is working with Support queues(AGI login user accounts) and meeting rooms. Voicemail system is not very good need some way to configure the boxes. They really need to redo this application for more standard settings. We have MOH working without any problems. Major down is no Graphical interface. No actual working manager. Got to get them to fix the Zombie lines. (I feel it's mainly do to our 390 and 452 phones. It works needs some fine tuning but it works. I have nothing good to say about the Aastra phones 390 or the 452. They are not really good for heavy use like we need! The Cisco 7960 is nice to look at but in the real world it's hard to get working and setup. If you don't know about Linux or are able to use scripts it's a real mess to keep up! This is where it's being held back as a real world player! This is the basic setup. Next step is outside offices connection. ___ What kind of resources are being consumed on the server? CPU,MEM,DISK,etc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP10S and Handset
Hi Robert, I haven't the HeadSet model but the lan switch model so I can't be of any help for you. Daniel rnc Info Lists a écrit: With some lively IP10S discussions here maybe someone knows about this issue: I can use the speaker phone ok. However the handset and switch hook do not seem to work. If I enable "headset" then I can get audio via the handset but still have to use the speaker phone button to take ot "off hook". Seems a bit wierd.. I have sent it to Swissvoice but no answer back yet. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 14:56, Ariel Batista wrote: > Ok here is a short paragraph on our use of Asterisk in the real world. Thank you Ariel - this is exactly the sort of information I am looking for :) I'm assuming this is actually for Avionica rather than you just writing from a different e-mail address? >From our own point of view, the lack of UI and dependency on scripting is no problem since we already have a Perl/PHP programming contingent here, so your words are actually encouraging :) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Gavin Hamill a écrit: On Tue, 2003-11-04 at 10:14, Daniel ANDRE wrote: Next I have left the context blank in my mcgp.conf and modified the default dialplan in my extensions.conf and now my IP10S can dial out. Hullo Daniel :) Can I request that you post the pertinent parts of your config to the list, since I'm sure I'm not the only one who would benefit from a set of known-working configs for these phones. I will make some clean-up in my files and post them in a day or two. I am not fully satisfied with my conf for now but it may help you. Daniel Personally, I'm on the verge of buying some SwissVoice handsets, simply because the mix of feature-set, price, and build quality seems to be untouchable. The GrandStreams are about the same price, but the build quality looks cheap and plastic - the IP10 actually looks like a business telephone. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
RE: [Asterisk-Users] 4 X100P's, 4 7960's Same Box?
7914's don't work with SIP Firmware. Just Skinny. * Supports SCCP but its very basic right now. bkw On Tue, 4 Nov 2003, David Gomillion wrote: > Cisco's CM can do the multiple line appearances, showing state. We > could not ever get ringback on any phone except the extension that it > was assigned to. > > Our workaround was to get a sidecar for each handset. With the side > cars, you get colored lights that blink different colors for different > call states. When it rang back, it would blink orange (I think) which > was a visual cue that it needed to be picked up. > > I have not yet tried this with *, as I just left BYU to go to a small > company with an annual IT budget of about a burger and 2 fries. But I > know the skinny protocol is going to support it. There may be some > features we need to fill out in the code for * to be able to take > advantage of those features. > > Hope this helps, > David Gomillion > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Brian West > Sent: Monday, November 03, 2003 10:10 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] 4 X100P's, 4 7960's Same Box? > > > I'm going to use Cisco 7960's for the phones; is there a better phone > I > > should be using? > > excellent choice. > > > I need to know if this is possible... > > > > On each phone program the appearances of 4 "Extensions" that are > really > > the 4 phone lines? > > yes for inbound. 1 rings 1... 2 rings 2.. and so on. Not the best > method > of doing this.. but ok. > > > If Phone "A" puts line 1 on hold can phone "B" see it on hold or will > it > > just show off hook? > > Not possible with SIP... might be with Skinny. Use parking or transfer. > > > Can the phones ring back (Even the one that did not put the call on > > hold) if it's not picked up in 30 or so seconds? > > Nope. (See previous answer) > > > If phone A is using line 1 and phone B is using line 2 will phone C > see > > the other 2 call appearances? > > Nope. (Not with SIP) > > > MOST IMPORTANT can I use 4 X100P's in the same box? What can I > expect? > > >From what I read in the posts the X100P is a $10 modem marked up to > pay > > for the Asterisk dev, and I don't mind paying for that I just want to > > make sure they are reliable and will get the job done, they will get a > > good workout for about 3 hours a day! > > I have 3 in my Dell Poweredge 500SC but you need to have a good bios. > One > that will let you put each card on its own IRQ. > > > This email transmission and any attachments are for the sole use of > the > > intended recipient(s) and may contain confidential and privileged > > information that is the sole property of PathAxis, Inc. Any > unauthorized > > review, use, disclosure or distribution is prohibited. If you are not > > the intended recipient, please contact the sender and destroy and > delete > > all copies of this email and any attachments. > > > These warnings are stupid. They should be at the top if you really want > people to read them. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call processing after a PIN
Hi folks, I would like some pointers to do a routine like this; 1 Call received at Asterisk 2 Asterisk answers with a voice prompt "enter a PIN number" 3 PIN received over DTMF 4 PIN is cross referenced in a database (SQL or FLAT file) 5 If a match is found, Asterisk prompt for "enter phone number" 6 Phone number is received over DTMF 7 Asteirsk route the call thanks in advance. Cheers Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
-- Original Message -- From: Ken Godee <[EMAIL PROTECTED]> >Ariel Batista wrote: >> Ok here is a short paragraph on our use of Asterisk in the real world. >> > ___ > >What kind of resources are being consumed on the server? >CPU,MEM,DISK,etc I am not a Linux person (Trying to learn) so I am not able to check this out! But I do have over 12 gig of disk space still available. If you have some program or setting I can run on the server to give me this info I would love to see it! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 2003-11-04 at 15:08, Steven Critchfield wrote: > I'm not trying to flame you for this message, but this is something that > is asked about quite often and doesn't exactly prove anything. Just > because I successfully pulled off an installation doesn't mean you will > be successful. No offence taken. Actually, I agree entirely. My intent is purely to show management that * is something worth allocating my time to research further, and not just some uber-geek project that's "fun, but of no commercial value." > Maybe what needs to happen to keep this question from coming up over and > over again would be to get this on the wiki. Someone want to create the > page and then post the link to that specific page to be filled in? Putting these on the Wiki isn't a bad idea, and is something I considered, but given the number of problems that people seem to have with it (e.g. it works fine, but is dead-slow from here...), and that e-mail reply takes fewer brain-cycles than registering for an acct, and formatting a Wiki posting, etc. People are doing the community a favour of providing the information, so it's in everyone's interests to make it as painless a process as possible. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the call manager. In my case, I have the message 'No available service' instead. What's wrong? Daniel -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multitech.
Hi All, I'm new to asterisk, can I use asterisk with a Multitech mvp 210? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
Check if you configured the clocking from their circuit correctly. You need to have span=1,1 ... in zaptel.conf Martin On Tue, 4 Nov 2003, Eduardo Goncalves wrote: > On Mon, 3 Nov 2003 17:15:21 -0600 > Don Pobanz <[EMAIL PROTECTED]> wrote: > > > > Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate > > > > start > > > > signaling), and just few seconds after this, all alarms are > > cleared. > > > > > > > > This problem ocurrs many times/day, and if are calls in > > > > progress, > > > > these calls just hang-up. > > > > Could it be an asterisk bug? Or may I contact the PSTN provider? > > > > > > I'd suggest your telco doing loopup and checking the circuit. > > > > > My telco checked the circuit last night and didn't find anything > wrong. > Now I'm lost. What should I check to find what's going on? > > > > Eduardo > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX2 Java library
On Tue, 2003-11-04 at 05:29, Alastair Maw wrote: > On 04/11/03 04:59, Steven Critchfield wrote: > > >>But C isn't as maintainable as nice Java apps, and it's as simple as > >>that. Basically, I'm after the most powerful interface possible to > >>Asterisk, but trying to make it as friendly as possible to code things > >>against. As far as our organization is concerned, that pretty much means > >>Java objects. > > > > So you bought that line of Marketecture didn't you. I think there are > > several large open source projects that prove that C is maintainable. > > Maintainability is really a function of organization. If you can't be > > organized, you will not produce very maintainable C code. > > Stop it right there. This mailing list isn't the place for language holy > wars. Java is the tool of choice for our organization, and therefore > Java code is more maintainable within it. Everyone here talks Java. Not > everyone talks C. If we hire less good programmers, Java hold their hand > and enforces structure in a way that C doesn't. I'm not arguing that > decent C code isn't maintainable. I'm stating that in our particular > case, Java fits the bill better. It's that simple. This isn't a language holy war. So far we have only pointed out how to get your needs done without needing to go and rebuild everything that is already working and good within asterisk in a language that was never meant to be a realtime language. We are steering you a direction where it should be easiest to accomplish in a reliable manner, and where you will actually be able to get help. If you go the Java/voip route, there won't be much if any help available from this group to help you out as you will be outside of the support here. > As for this: You needed to give proper attribution here. This is not a quote from me, but the way it is attributed above would make it seem this way. So to fix your goof, this quote was from rnc Info Lists <[EMAIL PROTECTED]> > > It all comes down to the number of CPU cycles needed to perform a > > given function. When doing real time processing, a few cycles here > > and a few there can add up to make a real difference. Object Oriented > > is nice for ease of writing/maintaining code but all of those objects > > have blocks of code behind them. A slight inefficiency there can > > really impact performance. Sure we have faster processors and lower > > cost memory every 6 months but thats no excuse for not writing the > > most efficient code possible. > > You're entirely wrong. Provided it's fast enough, it's all about > maintainability, extensibility and scalability. The bottom line is that > we're in it to make money. If it runs on hardware that's $100 cheaper, > but takes four times as long to develop, then total cost is much higher. > If it's easy to repurpose it at a later date, or easier to have someone > unfamiliar with the codebase to come along and customize it, that's > worth a lot. Both are significant, but maintainability is much more > important than performance, as that's where the real cost lies. The last statement is arguable. If you pay a few $k more per year for a decent programmer or potentially have to fill racks of colospace to put more hardware in place, you end up cheaper at the programmer. (BTW, I am NOT saying you aren't a decent programmer because you use Java. I wanted to head off that potential complaint before it starts.) We all know that C runs on the smallest of hardware without needing special hardware. Java only runs on small hardware when it has specialized silicon. But, that is purely an academic argument above as your company is already way to committed to the road it is on to be bothered to change. And even if it did consider the change, you have already laid out enough information to know you are not going to be able to do this inexpensively. So instead you will spend a fair amount of time outside of the communities expertise writing a customized application that only a very small amount would be even interesting to the group(IAX library). Ultimately though, I have doubts that there would be much interest in the Java IAX client as there is such a great number of C programmers here as witnessed by the couple of new IAX phone software that has shown up recently. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP10S and Handset
> Hi Robert, > > I haven't the HeadSet model but the lan switch model so I can't be of > any help for you. > > Daniel > I have the IP10S LAN Switch model too.. Thats why I find it wierd that the headset setting makes the difference ! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Steven Critchfield wrote: On Tue, 2003-11-04 at 03:48, Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case study, just a few lines to try and get a grip on the number of real installations. I'm not trying to flame you for this message, but this is something that is asked about quite often and doesn't exactly prove anything. Just because I successfully pulled off an installation doesn't mean you will be successful. There is obviously working systems out here otherwise there wouldn't be this much traffic on this list. Maybe what needs to happen to keep this question from coming up over and over again would be to get this on the wiki. Someone want to create the page and then post the link to that specific page to be filled in? Actually, this is the thing I find most usefull. There are not very many posts where the statement is about a fully functional asterisk system being deployed in a business. The vast majority of posts are about problems with asterisk, and in my case it has kept me from replacing our functional but aging PBX with asterisk. More success stories with details about the hardware and the environment would be of great benefit to this list, in my opinion. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call processing after a PIN
AGI will do it http://asterisk.gnuinter.net Panny - Original Message - From: "Sathya Weerasooriya" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 3:33 PM Subject: [Asterisk-Users] call processing after a PIN > Hi folks, > > I would like some pointers to do a routine like this; > > 1 Call received at Asterisk > 2 Asterisk answers with a voice prompt "enter a PIN number" > 3 PIN received over DTMF > 4 PIN is cross referenced in a database (SQL or FLAT file) > 5 If a match is found, Asterisk prompt for "enter phone number" > 6 Phone number is received over DTMF > 7 Asteirsk route the call > > thanks in advance. > > Cheers > > Sathya > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multitech.
Yes, we are just testing a MVP810 under SIP and it seems to work. Jorge Steve Bradwell wrote: Hi All, I'm new to asterisk, can I use asterisk with a Multitech mvp 210? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Ariel Batista wrote: -- Original Message -- From: Ken Godee <[EMAIL PROTECTED]> Ariel Batista wrote: Ok here is a short paragraph on our use of Asterisk in the real world. ___ What kind of resources are being consumed on the server? CPU,MEM,DISK,etc I am not a Linux person (Trying to learn) so I am not able to check this out! But I do have over 12 gig of disk space still available. If you have some program or setting I can run on the server to give me this info I would love to see it! ___ Quick and dirty from console prompt you can use "top" From a desktop you can try ie.. "xosview" Or install something like "gkrellm" http://www.gkrellm.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP10S and Handset
Hi! > I have the IP10S LAN Switch model too.. Thats why I find it wierd that > the headset setting makes the difference ! To me it looks like you can use the handset plug to insert a headset, at least that is what I understood from reading the user guide PDF. I even have come to think that the only different between 10 and 10s is the 2nd ethernet port, and that the 10 doesn't have any additional headset jacks (I am speculating though). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Tue, 4 Nov 2003 09:42:36 -0600 (CST) Martin Pycko <[EMAIL PROTECTED]> wrote: > Check if you configured the clocking from their circuit correctly. You > need to have span=1,1 ... in zaptel.conf > This is my zaptel.conf: span=1,1,0,cas,hdb3 alaw=1-8 e&m=1-8 loadzone = us defaultzone=us [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec?
Dan, the software crashes if you exit after hanging up. If exiting without doing anything first, it works OK. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Sent: Tuesday, November 04, 2003 5:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec? Hi Ray - Original Message - From: "Ray Burkholder" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 4:04 PM Subject: RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec? > Can anyone make any recommendations, from personal experience, on a good > softphone that has good look and feel, and of course reasonable sound > quality, and works with Asterisk? > > Ray Have you tried DIAX? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? Well, it's not technically a business, but we are running "*" for our research lab with the following hardware: 4 analog POTS lines ADTRAN TSU-600 FXO and FXS Plugins 20 GS budgetone 100, 2 Analog phones Asus A7V, 900MHz Athalon, 256 MB Ram, 60GB hard disk 100BaseT Ethernet environment using BayStack 450 switches Digium T100P RH9 APC Smart-Ups 700 Notes: The choice of the channel bank was based on the possibility of future expansion for handling several other labs (and the fact that I got is and 6-V.35 cards, which I', not using) for $99 on e-bay). The installed infrastructure for the phone system at our University dictated our choice of POTS lines. We were also not given any choice about the Baystack switches . The Budgetone phones are probably not the ideal solution for a larger installation (yet), but met our needs perfectly, perform very well and are easy to set up and install. The analog phones are to provide emergency service during power failures. The UPS is sized small because we have emergency power failover after about 30 seconds. The CPU and MOBO choices were based on experience with the products. A word of warning: Echo WILL be a potential problem on any system that has a transition point from POTS 2-wire lines to a TDM environment (see my previous posting). Here are some suggestions about dealing with it: 1) Use the highest possible CPU speed. Why: The single greatest problem with software echo cancellation is the computational intensity of the autocorrelation needed to determine the amount of echo at each discrete delay time. If the CPU is slow, the canceller will not be able to keep up with the real time nature of the computations. 2) Enable MMX options (assuming that you have a CPU which supports it). Why: Same as in item 1. In my case, this provided the single greatest improvement. 3) Lower the number of taps in the echo canceller. Why: While this may seem counter intuitive, since the number of taps determines the amount of delay the canceller can deal with, most of the time the delay you hear is much longer than the delay seen by the echo canceller. Decreasing the number of taps reduces the size of the autocorrelation and reduces the comptational load. This allows the canceller to learn faster and keep up more easily. 4) Turn off the KDE, etc. Why: Saves computation cycles for the echo cancellation. This is no joke, you can get on the phone, start up the desktop and hear the echo return. 5) Attempt to balance the hybrid at the 2-line to 4 line interface. Why: 99% of the time, this is where the echo originates and this is where is should be fixed. Unfortunately, this is not for the faint of heart, but if your line card has a hybrid balance adjustment (many don't), use it. Also, with multiple simultaneous calls, this may be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this myself, but you might try something as simple as a 500 ohm variable resistor in series with the ring line and adjust for minimum echo. If it gets worse, you haven't lost anything, just take the resistor out of the line. If it works, measure the value of the resistor when set for minimum echo and replace it with a fixed value resistor. 6) Try messing with Tx and Rx gains. Why: This is a reincarnation of the technique used to cancel echo on long lines in the early part of the 20th century. The idea is that the perception of echo gets worse as echo volume increases and as delay increases. You can sometimes reduce echo to an "acceptable" level if you attenuate it enough, especially if the echo canceller takes care of part of it. The problem is that you will likely run into unacceptably low volume levels. That's why this technique was abandoned by the PSTN: You've all seen the movies of people in the 1920's yelling into phones on long distance calls! Finally, while I had to do some head scratching and a lot of reading to get the system set up, I would have to say that the installation went without any major problems. Once configured, it runs flawlessly and requires very little maintainence. Steve Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Help with SIP/H323.
Hi list,
why I cannot hear
voice when I call from a SIP telephone
(Budgetone and others) to a H323 telephone (several models)?
could anybody please give any
idea to solve this issue?
Please, let me know.
Thanks in Advance.
N.B.
The configuration for "extensions.conf",
"sip.conf" and "h323.conf" files are:
***
extensions.conf
***
[default]
.
.
[outgoing]
exten=>_7XX,1,Goto(voip-h323|${EXTEN}|1)exten=>_*XX,1,Goto(servicios|${EXTEN}|1)
exten=>_X,1,Dial(Zap/1/${EXTEN}|30)exten=>_X,2,Playback(invalid)exten=>_X,3,Hungup()
exten=>_X,1,Playback(invalid)exten=>_X,2,Hungupexten=>_XX,1,Playback(invalid)exten=>_XX,2,Hungupexten=>_,1,Playback(invalid)exten=>_,2,Hungupexten=>_X,1,Playback(invalid)exten=>_X,2,Hungupexten=>_XX,1,Playback(invalid)exten=>_XX,2,Hungupexten=>_XXX,1,Playback(invalid)exten=>_XXX,2,Hungupexten=>_,1,Playback(invalid)exten=>_,2,Hungup
exten=>i,1,Playback(invalid)exten=>t,1,Hungup
[voip-h323]
;
SIP::exten=>701,1,Dial(SIP/701)
;
SIP::exten=>702,1,Dial(SIP/702)
;
H323::exten=>703,1,Agi(AceptaLlamada.php)exten=>703,2,Dial(h323/3|17|tTm)exten=>703,3,VoiceMail(u703)exten=>703,103,VoiceMail(b703)
;
H323::exten=>710,1,Agi(AceptaLlamada.php)exten=>710,2,Dial(h323/10|17|tTm)exten=>710,3,VoiceMail(u710)exten=>710,103,VoiceMail(b710)
..
*
sip.conf
[general]port = 5060 ; Port to
bind tobindaddr = 0.0.0.0 ; Address to bind tocontext =
outgoing ; Default for incoming callssrvlookup = yes ;
Enable SRV lookups on outbound calls;pedantic = yes ;
Enable slow, pedantic checking for
Pingteltos=lowdelay;tos=184;maxexpirey=3600 ; Max length
of incoming registration we allow;defaultexpirey=120 ; Default
length of incoming/outoing registration;notifymimetype=text/plain ;
Allow overriding of mime type in NOTIFY;videosupport=yes ; Turn
on support for SIP videodisallow=all ; Disallow all
codecsallow=alawallow=ulaw ; Allow codecs in order of
preference;allow=ilbc
[701]type=friendusername=701fromuser=701secret=701host=dynamicdefaultip=192.168.0.151;mailbox=701context=outgoingcanreinvite=yesdtmfmode=infocallgroup=1pickupgroup=1
[702]type=friendusername=702fromuser=702secret=702host=dynamicdefaultip=192.168.0.152mailbox=702context=outgoingcanreinvite=yesdtmfmode=infocallgroup=1pickupgroup=1
.
***
h323.conf
***
[general]port = 1720bindaddr =
0.0.0.0tos=lowdelay;
amaflags=billing;
disallow=all
; turns on all installed
codecs;disallow=g723.1
; Hm... Proprietary, don't use
it...allow=gsm
; Always allow GSM, it's cool
:);allow=ulawallow=alaw;allow=g729
;
noFastStart=yesnoH245Tunneling=yesnoSilenceSuppression=yes;jitter=20;dtmfmode=inband;
gatekeeper = 192.168.0.207;AllowGKRouted =
yes;context=outgoing;
[CAC-IP] ;our
computer.type=h323prefix=9,7,*,8context=outgoing;
RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec?
Same here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 4 Nov 2003, Stephen R. Besch wrote: > 5) Attempt to balance the hybrid at the 2-line to 4 line interface. > Why: 99% of the time, this is where the echo originates and > this is where is should be fixed. Unfortunately, this is not for the > faint of heart, but if your line card has a hybrid balance adjustment > (many don't), use it. Also, with multiple simultaneous calls, this may > be the only real solution. Part of the problem arises from the use of > lower impedance telephone wiring nowdays. The typical characteristic > impedance of Cat5 twisted pair is about 100 ohms and many line cards are > optimized for a 600 ohm line. This is made worse if the DC resistance of > the wiring to the CO switch is relatively low. I haven't tried this > myself, but you might try something as simple as a 500 ohm variable > resistor in series with the ring line and adjust for minimum echo. If > it gets worse, you haven't lost anything, just take the resistor out of > the line. If it works, measure the value of the resistor when set for > minimum echo and replace it with a fixed value resistor. I've had to do similar things to lower loop current. Be sure you get at least 1 watt resistors and put the same size on the tip and ring. Putting resistance on only one wire will throw other things off. dave -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec?
Hi, - Original Message - From: "Shoval Tom" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 04, 2003 5:58 PM Subject: RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec? > Dan, the software crashes if you exit after hanging up. > If exiting without doing anything first, it works OK. > I'll solve this issue (present only on some systems) in the next release (0.9.2), together with some other discovered bugs. Keep on eye on this list. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why does context order make a difference in IAX.CONF
so why does this work properly [iax-ll] type=friend context=iax-ll-in trunk=yes host=iax2.ll.com [iax-] type=friend auth=md5 secret=blahblah context=iax--in trunk=yes host=rx8..net and this doesn't [iax-] type=friend auth=md5 secret=blahblah context=iax--in trunk=yes host=rx8..net [iax-ll] type=friend context=iax-ll-in trunk=yes host=iax2.ll.com What happens is a call that comes in over IAX- gets dropped into the iax-l-in context if iax- is first in iax.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring to Meetme
Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can subsequently transfer it to the meetme? All of this must be done through the management interface and not require users pushing any button.s Thanks. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
