[Asterisk-Users] NewB: Cisco 7910
Dear Sirs & Madams, Has anybody connected and sucesffully used a Cisco 7910 to an * PBX system? Also, how hard was it to setup? Any assistance would be greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues
I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent "agent 1001 hang up on customers. they must be pissed off". I agreed. My queues.conf file: [agents] ackcall=no agent => 1001,1001,xx ss My queues.conf file: [incoming] announce = incoming strategy=ringall musice = default member => Agent/1001 member => Agent/1002 My extensions.conf : exten => 28,1,AgentCallbackLogin(|@local) exten => 29,1,Queue(incoming) In order to annonce to agent the correct queue does it have to have a gsm file to playback the name of the queue ie "incoming" in this case? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
RE: [Asterisk-Users] OT Superbowl = Linux Shake up to the world..
http://www-306.ibm.com/e-business/doc/content/lp/prodigy.html?P_Site=S90 Linux, the Future is Open. An IBM Commercial shown with the Child Prodigy, it's not the first time they've shown it. - Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Sunday, February 01, 2004 11:28 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] OT Superbowl = Linux Shake up to the world.. [EMAIL PROTECTED] <> wrote: > Linux, Shake up the world > > oops sorry, > > test... testing 123 > [EMAIL PROTECTED] <> wrote: > I laughed out loud, and then looked around at all the other people in > the room who were staring at me because they didn't understand the > significance of the statement. For those who haven't seen the advert (I assume this is about an ad played at the superbowl) could you perhaps include a little more detail... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT Superbowl = Linux Shake up to the world..
[EMAIL PROTECTED] <> wrote: > Linux, Shake up the world > > oops sorry, > > test... testing 123 > [EMAIL PROTECTED] <> wrote: > I laughed out loud, and then looked around at all the other people in > the room who were staring at me because they didn't understand the > significance of the statement. For those who haven't seen the advert (I assume this is about an ad played at the superbowl) could you perhaps include a little more detail... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?
Title: Message Yeah, but without a sound card; they won't work. My suggestion would be to place a call from an outside line (or cell phone) through the * to voicemail or the demos to prove that the system works. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark HaglerSent: Sunday, February 01, 2004 10:06 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0? Use a soft phone as an endpoint. There are a variety of SIP and IAX softphones you can use to place a call through your Asterisk box over IP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael ZhengSent: Sunday, February 01, 2004 7:34 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0? Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best,Michael Do you Yahoo!?Yahoo! SiteBuilder - Free web site building tool. Try it! --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.572 / Virus Database: 362 - Release Date: 1/27/2004
RE: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?
Use a soft phone as an endpoint. There are a variety of SIP and IAX softphones you can use to place a call through your Asterisk box over IP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Zheng Sent: Sunday, February 01, 2004 7:34 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0? Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best, Michael Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
RE: [Asterisk-Users] Superbowl = Linux Shake up to the world..
I laughed out loud, and then looked around at all the other people in the room who were staring at me because they didn't understand the significance of the statement. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Cardenas Sent: Sunday, February 01, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Superbowl = Linux Shake up to the world.. Linux, Shake up the world oops sorry, test... testing 123 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.572 / Virus Database: 362 - Release Date: 1/27/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme without zaptel hardware
Has anyone had any success using the ztdummy module and doing meetme/conferencing with out zaptel hardware installed? Paul
Re: [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?
At 10:34 PM -0500 2/1/04, John Todd wrote: At 3:09 PM -0600 2/1/04, Michael Graves wrote: Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] "Kick at the darkness 'till it bleeds daylight" - Bruce Cockburn No, but if you could maybe eval one and put up a review like Adam Hart did for the Mediatrix, it would be a big benefit back to the community. JT Ah, crap. I have mis-quoted. The nice review on the Mediatrix was by Rich Adamson, not Adam Hart. My apologies. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?
At 3:09 PM -0600 2/1/04, Michael Graves wrote: Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] "Kick at the darkness 'till it bleeds daylight" - Bruce Cockburn No, but if you could maybe eval one and put up a review like Adam Hart did for the Mediatrix, it would be a big benefit back to the community. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?
Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best,Michael Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
[Asterisk-Users] Re: Adtran 750 DID question.
Scott, You can't see DNIS on any channels/Line? If I understand correctly you can't see digits coming in from your Telco? And it falls into your S context? Try changing your Adtran Firmware, I tried on L35 and L36 it didn't worked properly, so I am using L34. Also, try changing the wink time in Zapata.h from 150MS to 250MS, once you change the timing you have to recompile Zapata. Kd Subject: RE: [Asterisk-Users] Re: Adtran 750 DID question. Date: Fri, 30 Jan 2004 19:16:26 -0500 From: "Bisker, Scott (7805)" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Yes. immediate=3Dno is in zapata.conf before the channel declaration. = This makes absolutely no sense at all. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Superbowl = Linux Shake up to the world..
Linux, Shake up the world oops sorry, test... testing 123
Re: [Asterisk-Users] Compiling while * is running
On Sun, Feb 01, 2004 at 04:51:30PM -0600, Steven Critchfield wrote: > > This isn't intended as a flame bait. The original message should have > been more clear that I thought you where experiencing crap in windows. Heh. I haven't used windows since 1995 :) In fact, with HP-UX you cannot delete or rename or overwrite a shared library if it is in use, so you would *have* to stop the process before doing a "make install". For example, http://web.gat.com/comp/analysis/mdsplus/textfilebusy.html Talks about this phenomenon. > How the hell did HP-UX get trusted status for military use if that is > true? HP was/is a big military contractor long before HP-UX came into being, so perhaps that has something to do with it... /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?
As far I have reasearched none of the FXO devices were perfect except Cisco VIC ones. If you are looking for reliability I recommend not to use those ones except cisco. Kannaiyan - Original Message - From: "Michael Graves" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, February 01, 2004 9:09 PM Subject: [Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters? > Anyone here have experience with these devices? They would ppear to be > an affordable alternative to multiple X100Ps. > > Michael > > -- > Michael Graves [EMAIL PROTECTED] > Sr. Product Specialist www.pixelpower.com > Pixel Power Inc. [EMAIL PROTECTED] > > "Kick at the darkness 'till it bleeds daylight" - Bruce Cockburn > > ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNIS on X100P
Hello, I have 6 analog lines that ring down coming into my office that support DNIS, my current phone system (SRX) displays the "called number" on the screen of the operator phone, IE; xxx-7873 = Netxn, xxx-7874 = Dolphinsafe, etc. Does asterisk support any type of features to distinguish between the numbers dialed? Thanks! Chris Wilson
Re: [Asterisk-Users] Dial via sip gateway?
> > From what I can tell (box is about 48 hrs old for me), it > > seems to be a rather incomplete or just-bare-sip-minimum > > functionality. It also appears as though all four ports are > > treated as a group-of-lines, and one doesn't have any choice > > (from a sip perspective) on which port to use for outgoing > > calls. Since this one is set up with 1:home, 2:business, > > 3:outgoing calls I really need to be able to select which > > port * is going to use, particularly since outgoing 'home' > > long distance calls must use a different port then for > > outgoing 'business' calls. > > I have an idea of a crude hack that just might work - e.g. if you need > to dial a number on line 3, first make two outgoing calls to a bogus > number (just to keep the lines busy for a second) and then place the > 3rd call to the destination you want - if I understand the situation > correctly, the 1204 should dial on the 3rd line then and the first two > calls should drop quickly (no such number). Of course, in that case > you need to keep the line state e.g. in the DB so that, say, line 1 in > use doesn't mess things up. > > Yes, I know it's ugly. If it's also bound not to work, I'm all ears as > to *why* :) Yup, that's a very ugly one. Given this is an eval box with an option to buy, I'd rather send it back. Other then the register function, the box appears to be a very nice one. Maybe a little pricey, but would bet it fits into a very large number of businesses/homes very nicely. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
On Sun, 2004-02-01 at 16:38, William Waites wrote: > On Sun, Feb 01, 2004 at 04:21:23PM -0600, Steven Critchfield wrote: > > > > Dude maybe you need to learn more Unix programing and leave those toy > > OSes alone. Once a module is loaded, there should be no need to read the > > version on the file system again. Your problem would be loading new > > modules into a running version where there may have been an api change. > > Steven, stop flame-baiting. HP-UX, for example, might be an > ugly proprietary SysV monster, but it's far from a toy. > > There do exist broken dynamic loader implementations based > on mmap(2). This isn't intended as a flame bait. The original message should have been more clear that I thought you where experiencing crap in windows. How the hell did HP-UX get trusted status for military use if that is true? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
On Sun, Feb 01, 2004 at 04:21:23PM -0600, Steven Critchfield wrote: > > Dude maybe you need to learn more Unix programing and leave those toy > OSes alone. Once a module is loaded, there should be no need to read the > version on the file system again. Your problem would be loading new > modules into a running version where there may have been an api change. Steven, stop flame-baiting. HP-UX, for example, might be an ugly proprietary SysV monster, but it's far from a toy. There do exist broken dynamic loader implementations based on mmap(2). /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling while * is running
On Sat, 2004-01-31 at 20:02, William Waites wrote: > On Sat, Jan 31, 2004 at 07:43:46PM -0600, Brian West wrote: > > Nope I do make install all the time with asterisk running without ONE > > problem. > > As I said, this behaviour is specific to some implementations > of dynamic loadable modules. It depends what OS (and in some > cases what version of the OS) you are running. Dude maybe you need to learn more Unix programing and leave those toy OSes alone. Once a module is loaded, there should be no need to read the version on the file system again. Your problem would be loading new modules into a running version where there may have been an api change. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can a variable be redefined within extensions.conf
Hi
> Can I define a variable in globals like this:
>
> [globals]
> timeout=60
>
> and then in another context, redefine that same variable and only have
> the new value affect the call that hit that particular extension ?
yes, works this way.
mind : each var is unique for each call... a global
var is simply shared between contexts, not between calls.
each call is a standalone entity
>
> [example]
> exten => _9NXX,1,DBget(blah/blah)
> exten => _9NXX,102,Goto(3)
> exten => _9NXX,2,SetVar(#timeout=20)
> exten => _9NXX,3,Dial(${PSTN},${EXTEN:1},${timeout})
> exten => _9NXX,104,Do something if busy
> exten => _9NXX,4,Do something else if no answer
>
> So, if a call goes priority 1,102,3... ${timeout} would = 60 but if the
> call went priority 1,2,3, then ${timeout} would =20.
yes.
>
> Or, could I just leave the variable definition out of globals all
> together ? What would ${timeout} evaluate to if it isn't defined ? I
> suspect dial wouldn't be happy since the comma right before timeout
> would still be there
so this way dial isn't unhappy... simply dial will "dial"
with no timeout
Matteo.
--
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl
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Re: [Asterisk-Users] Dial via sip gateway?
> > I don't believe the above will work. There is only one IP address for > > the box, and no way that I've found to send a sip packet to the box with > > "additional" information that would suggest using port 1 vs port 2. From > > what others have hinted at (and it seems the majority of us are limited > > either to what's printed or experimentation), the 1204 has an internal > > function that kind of resembles a trunk group. "It" decides which port > > to use. > > > > As mentioned previously, the sip "register" function in the box is inop > > in both directions, therefore there does not seem to be a way to address > > the ports through contexts or anything else. Mediatrix has provided the > > mib variables where one can enter a different password for each port, > > but that has no value either since the register function doesn't work. > > What happens if you don't use a register => line in sip.conf, but do > include a section like: > [mediatrixport1] > username= > password= > host= The above is basically what I did, however since the 1204 never attempts to register, the username and password have no value. The host= is the only statement above that has value, and its the "only" thing that can be used to associate a context with the gateway. Attempts to use a register statement within * (and watching packets with a sniffer), the register attempt is greated with "501 Not Implemented" from the 1204. > Just to check my theory, I did some testing via fwd. I discovered that if > I include a register => line with my fwd info, then when I call my fwd > number (outbound through iaxtel) it rings in. But I can't call out via > fwd. So then I put in my [fwd] service definition, removed the register > line, and waited for the old registration to expire. Then I tried calling > my fwd number (again through iaxtel). This time I got the message about > the user being offline. But now I can call out via fwd, even though calls > wouldn't come in. This demonstrates that the [fwd] section is used by > Dial() when I try to place a call out through that service, and that the > register line isn't needed for the outbound call. Sure, but fwd and your asterisk both understand the register function. The 1204 does not. > Somebody mentioned that the mediatrix lets you set a unique > username/password for each of its ports. That was me that said it in an earlier email attempting to find out if it was "me" or the "1204" that didn't understand what was going on. Turned out to be the 1204. > It seems that you could set up > four service definitions, each using a different user/pwd pair. Then * > will use a different user/pwd pair to log in to the mediatrix, depending > upon which service definition was called for by the Dial() statement. which, again, all depends on the register function working. > Or does the mediatrix not really have a distinct user/pwd pair for > accessing each port? It has the mib variables and one can set them, the 1204 just doesn't do anything with them. The bottom line really is "501 Not Implemented", period. Until that's implemented there really isn't anyway to address individual ports in any form that is reasonable. For what its worth, it would appear from the Mediatrix web site (takes a little digging) the group behind writing the sip code must have had some financial problems. They received some funding in November along with apparently some senior management changes. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?
[EMAIL PROTECTED] wrote: I'm trying to set up an Asterisk system for a small office, and one thing I haven't figured out yet is how to best provide reliability. One way to go seems to be a T1 for all my incoming phone lines. What if that T1 goes down? Can I use mutiple POTS lines in conjuction with a T1, all connecting to my Asterisk server? What if my Asterisk server fails? Should I use two Asterisk servers, one connected to the T1 and one to the POTS lines? Thanks, P. Its all come down to how critical the telephone service is to your business, and how much money you have.. I would say that if you are that worried about your Telco not providing a reliable service than you should maybe look at using another Telco.. After that its just redundancy, doubling up on everything.. Hard drives running RAID and dual power supplies, maybe a complete spare server to plug in if the first fails.. At the end of the day you need to decied what uptime you REQUIRE and how much money you have to spend and try and get the two as close together as possible.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can a variable be redefined within extensions.conf
Can I define a variable in globals like this:
[globals]
timeout=60
and then in another context, redefine that same variable and only have
the new value affect the call that hit that particular extension ?
[example]
exten => _9NXX,1,DBget(blah/blah)
exten => _9NXX,102,Goto(3)
exten => _9NXX,2,SetVar(#timeout=20)
exten => _9NXX,3,Dial(${PSTN},${EXTEN:1},${timeout})
exten => _9NXX,104,Do something if busy
exten => _9NXX,4,Do something else if no answer
So, if a call goes priority 1,102,3... ${timeout} would = 60 but if the
call went priority 1,2,3, then ${timeout} would =20.
Or, could I just leave the variable definition out of globals all
together ? What would ${timeout} evaluate to if it isn't defined ? I
suspect dial wouldn't be happy since the comma right before timeout
would still be there
Thanks
--Lance
Go Panthers :)
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[Asterisk-Users] Luxoncomm 3800 series FXO/FXS adapters?
Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] "Kick at the darkness 'till it bleeds daylight" - Bruce Cockburn ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004 08:21:55 -0600, Rich Adamson wrote [long snip] > No, the manual is very verbose but no * examples at all. The > box sells as either a 323 or sip, with different images > (sort of like C7960's) and different manuals. > > The box does not support the "register" function in either > direction. I just tried the * sip register, and got a "501 > Not Implemented" with sniffer. > > From what I can tell (box is about 48 hrs old for me), it > seems to be a rather incomplete or just-bare-sip-minimum > functionality. It also appears as though all four ports are > treated as a group-of-lines, and one doesn't have any choice > (from a sip perspective) on which port to use for outgoing > calls. Since this one is set up with 1:home, 2:business, > 3:outgoing calls I really need to be able to select which > port * is going to use, particularly since outgoing 'home' > long distance calls must use a different port then for > outgoing 'business' calls. I have an idea of a crude hack that just might work - e.g. if you need to dial a number on line 3, first make two outgoing calls to a bogus number (just to keep the lines busy for a second) and then place the 3rd call to the destination you want - if I understand the situation correctly, the 1204 should dial on the 3rd line then and the first two calls should drop quickly (no such number). Of course, in that case you need to keep the line state e.g. in the DB so that, say, line 1 in use doesn't mess things up. Yes, I know it's ugly. If it's also bound not to work, I'm all ears as to *why* :) > > The entire box (4 ports) has only a single IP, so if the > dial sip command doesn't have any additional > parameters/strings to destinguish selected ports, guess I'll > return it to the reseller. There appears to be a way to set certain > types of filters on a per port basis in the box, but I can't > see how that could be used to differentiate home vs business > calls, etc. > > Since I don't know anything about 323, does that control > protocol allow some sort of sub-selection where each port > would be addressable? If not, it certainly seems as though > Mediatrix needs to go back to work on their code or something. > > Can you think of any other way that * might interact with > this thing via sip? > > Rich > Regards, Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I provide redundancy and reliability w/ Asterisk?
I'm trying to set up an Asterisk system for a small office, and one thing I haven't figured out yet is how to best provide reliability. One way to go seems to be a T1 for all my incoming phone lines. What if that T1 goes down? Can I use mutiple POTS lines in conjuction with a T1, all connecting to my Asterisk server? What if my Asterisk server fails? Should I use two Asterisk servers, one connected to the T1 and one to the POTS lines? Thanks, P. -- Philip J. Hollenback [EMAIL PROTECTED] http://www.hollenback.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004, Rich Adamson wrote: > I don't believe the above will work. There is only one IP address for > the box, and no way that I've found to send a sip packet to the box with > "additional" information that would suggest using port 1 vs port 2. From > what others have hinted at (and it seems the majority of us are limited > either to what's printed or experimentation), the 1204 has an internal > function that kind of resembles a trunk group. "It" decides which port > to use. > > As mentioned previously, the sip "register" function in the box is inop > in both directions, therefore there does not seem to be a way to address > the ports through contexts or anything else. Mediatrix has provided the > mib variables where one can enter a different password for each port, > but that has no value either since the register function doesn't work. What happens if you don't use a register => line in sip.conf, but do include a section like: [mediatrixport1] username= password= host= Just to check my theory, I did some testing via fwd. I discovered that if I include a register => line with my fwd info, then when I call my fwd number (outbound through iaxtel) it rings in. But I can't call out via fwd. So then I put in my [fwd] service definition, removed the register line, and waited for the old registration to expire. Then I tried calling my fwd number (again through iaxtel). This time I got the message about the user being offline. But now I can call out via fwd, even though calls wouldn't come in. This demonstrates that the [fwd] section is used by Dial() when I try to place a call out through that service, and that the register line isn't needed for the outbound call. Somebody mentioned that the mediatrix lets you set a unique username/password for each of its ports. It seems that you could set up four service definitions, each using a different user/pwd pair. Then * will use a different user/pwd pair to log in to the mediatrix, depending upon which service definition was called for by the Dial() statement. Or does the mediatrix not really have a distinct user/pwd pair for accessing each port? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review
Rich Adamson wrote: Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. Trouble shooting is limited to the SNMP manager only. The manager can be used to view configuration data, however needed dynamic operational statistics are limited to mib2 definitions only. For example, when trying to determine the souce of choppy MOH sound, I wanted to check the Ethernet port speed. There was no mib variable defined for this purpose. I found the syslog feature pretty niffty. You crank the syslog up to level 5 and get a lot of info. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
Greg,
> > So try something like this in extensions.conf:
> > exten => 101,1,Dial(SIP/@mediatrixport1)
> > exten => 102,1,Dial(SIP/@mediatrixport2)
> > exten => 103,1,Dial(SIP/@mediatrixport3)
> > exten => 104,1,Dial(SIP/@mediatrixport4)
>
> Oops, maybe I should have written these extensions to be more like this:
> exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _8NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _7NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _6NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
>
> so that you can choose which port you'll dial out on by prefixing your
> number with 9/8/7/6.
I don't believe the above will work. There is only one IP address for the box,
and no way that I've found to send a sip packet to the box with "additional"
information that would suggest using port 1 vs port 2. From what others have
hinted at (and it seems the majority of us are limited either to what's printed
or experimentation), the 1204 has an internal function that kind of resembles
a trunk group. "It" decides which port to use.
As mentioned previously, the sip "register" function in the box is inop in
both directions, therefore there does not seem to be a way to address the
ports through contexts or anything else. Mediatrix has provided the mib
variables where one can enter a different password for each port, but that
has no value either since the register function doesn't work.
You've sort of touched on a method that might work, by prefixing called numbers
with a digit, then strip it in the 1204, etc. However, when you think that
process through for anything other than the simpliest of cases, it creates
a fairly major dialplan management issue. For the price of the box, think
I'll delay the purchase for now.
Rich
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[Asterisk-Users] Configuring Firefly Network in *
I did get it to work, and can place and receive calls through the Firefly
network via *.
Compared to iaxtel or FWD, there is a significantly higher amount of
latency, but it is workable.
For some reason, this needed to be the last entry in my iax.conf or it
would try to authenticate with a different user ID when receiving calls
(and obviously would fail.
Relevant section from my iax.conf:
register => 87210384:[EMAIL PROTECTED]
[87210384]
context = firefly-in
secret=xxx
auth=md5
type=friend
username=87210384
host=firefly.virbiage.com
qualify=yes
trunk=no
In my dialplan:
[globals]
...
FIREFLY=IAX2/[EMAIL PROTECTED]
[fireflycalls]
; Firefly Calls
;
exten => _**.,1,SetCallerID(87210384)
exten => _**.,2,SetCIDName(Joel Maslak)
exten => _**.,3,Dial(${FIREFLY}/${EXTEN:2},,)
exten => _**.,4,Playback(invalid)
[good-user]
include => extensions
include => extensions-services
include => good-outbound
include => fireflycalls
For inbound:
[firefly-in]
exten => s,1,GotoIfTime(07:00-21:00,mon-fri,*,*?incoming,s,1)
exten => s,2,GotoIfTime(09:30-21:00,*,*,*?incoming,s,1)
exten => s,3,VoiceMail(u10)
Where "incoming" is my standard incoming context. Calls come in without
any DNID, they go straight into the "s" extension in the firefly-in
context. The time restrictions are my standard "non-PSTN annoyance at
2:00 AM filter".
With this config, I can dial Firefly users by dialing ** + their Firefly
number.
--
Joel Maslak
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Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004, Greg Hill wrote:
> So try something like this in extensions.conf:
> exten => 101,1,Dial(SIP/@mediatrixport1)
> exten => 102,1,Dial(SIP/@mediatrixport2)
> exten => 103,1,Dial(SIP/@mediatrixport3)
> exten => 104,1,Dial(SIP/@mediatrixport4)
Oops, maybe I should have written these extensions to be more like this:
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _8NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _7NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _6NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
so that you can choose which port you'll dial out on by prefixing your
number with 9/8/7/6.
Greg
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[Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review
Product Review
Mediatrix 1204 4-Port SIP FXO Gateway
Firmware: v2.4.10.69 - US Version
US Retail: ~$750, Street Price: ~$450.
The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks
and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn
lines in either Loop Start or Ground Start mode, handles incoming CallerID,
and generates either Dial Tone (back towards the incoming pstn caller) or
redirects the call to a specific pre-programmed sip proxy extension. The
1204 can only be programmed through an SNMP (Simple Network Management
Protocol) manager. A Windows-based SNMP manager is supplied with the unit;
but no Unix-based manager. (And, no telnet, no web.)
To use the 1204 with Asterisk, each of the four pstn lines "must" be redirected
to an Asterisk extension. In this eval case, port 1 was redirected to x3091,
port 2 to x3092, etc. The 1204 detects the incoming call, and about midway
through the second ring, sends a sip Invite with the CallerID (if available)
to the defined sip proxy server (Asterisk). (After Asterisk completes the call
to another sip phone and the pstn caller hangs up, the Asterisk sip phone
will continue to ring for at least two-to-four additional ringing cycles.)
The firmware version tested did not support the sip "register" function even
though parameters were provided to enter the IP address of a registrar. As
a result, no userid/passwords or other security features are available.
Sip.conf security entries are limited to "host=" and context=
". All incoming port 1 calls are directed to an extension.conf
construct similar to exten => 3091,1,Goto(my-ivr) contained within the
section.
Again since the 1204 does not support the sip "register" function, outgoing
pstn calls from Asterisk can only be sent to the 1204 with commands similar to:
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
where the 1204 selects one of the non-busy four pstn ports at random to initiate
the pstn call. The reviewer could not find a way to direct specific Asterisk
calls to specific 1204 ports, and believes Mediatrix needs to fully implement
the "register" command on a per-port basis to aid in this requirement.
Echo cancellation and transmission levels were excellent on all inbound and
outbound calls. Ringback tone and Music on Hold (MOH) were extremely choppy
until the Port1DspVoiceActivityDetection = 0 parameter disabled this function.
NAT is supported according to the documentation, however I did not test this
to see if it actually worked. The standard rtp redirection (canreinvite=yes)
appeared to function properly.
As mentioned, the only way to configure the 1204 is via an SNMP Manager. There
is no way to change/secure the SNMP-v1 community string, therefore this box
should never be exposed to the Internet. The *.pdf documentation files are
very verbose and good (Admin = 196 pages); however there are no references to
Asterisk, leaving the reader to guess at how some functions actually
inter-operate, etc.
Opinion:
It would appear the 1204 is oriented to inter-operate with another 1204 across
the Internet, creating essentially a virtual pstn line extension to some
distant point. The box is available with either H-323 or SIP images, but not
both. One can only assume the incomplete SIP implementation is the result of
retrofitting the 323-based box into the SIP world. Since much of the *.pdf
documentation and files were dated March/April 2003, it does not appear
that SIP advancement is high on Mediatrix's list of priorities. Support for
the unit is limited by Mediatrix to "resellers only", therefore obtaining
any relevant support data in a timely manner is 100% dependent on how well
your reseller will support you.
Trouble shooting is limited to the SNMP manager only. The manager can be used
to view configuration data, however needed dynamic operational statistics are
limited to mib2 definitions only. For example, when trying to determine the
souce of choppy MOH sound, I wanted to check the Ethernet port speed. There
was no mib variable defined for this purpose.
Overall, the 1204 functioned very well for what has been implemented, however
a more complete sip implementation, better technical support, and limited
trouble shooting access will delay my decision to purchase this unit.
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Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004, Rich Adamson wrote: > The above does not seem to work either. Since the mediatrix has four pstn > ports, there must be a way to construct a Dial command that would embed > a userid:password, port alias name, or something like that. Just can't find > any reference to what that syntax would look like. (The gateway is properly > handling incoming pstn calls, just not the outgoing pstn attempts.) > > Really need to the sip dial command to include... > - the string of digits to be called > - either a userid:password, or, port alias name (or both) > - ip address of the gateway > > Anybody have a clue what that dial sip syntax would look like I have only recently begun actually playing with *, but I'll venture a guess.. You (or somebody else) mentioned that you can force a call to go out a particular port on the Mediatrix by using the username/password pair which corresponds to that port, and this guess is based on that assumption. (I hope it's a valid assumption!) At http://www.voip-info.org/wiki-Asterisk+SIP+channels, under "Using a SIP channel in extensions.conf," we read that the dial string format is either SIP/@ or SIP/peer/exten. may be a hostname of a SIP proxy server, a domain where * should look for a SRV record, or a service defined in sip.conf. So try something like this in extensions.conf: exten => 101,1,Dial(SIP/@mediatrixport1) exten => 102,1,Dial(SIP/@mediatrixport2) exten => 103,1,Dial(SIP/@mediatrixport3) exten => 104,1,Dial(SIP/@mediatrixport4) and then define those services in sip.conf: [mediatrixport1] username= password= host= [mediatrixport2] username= password= host= and so on for ports 3 and 4. I think a setup like this will allow you to use distinct username/password pairs for connections to the same SIP proxy. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
I haven't taken the time to reverse engineer this on * but subscribe is used in SIP for serveral things: 1. Message Waiting Indicator (MWI). Asterisk seems to send out a NOTIFY even with no SUBSCRIBE though. :) 2. SIMPLE (SIP Instant Message & Presence Leverage Extensions). The SUBSCRIBE/NOTIFY is used for the "presence" portion of this. The Windows Messenger and MSN Messenger 4.6/4.7 support a pseudo-standard form of this. We use this our development for "phone" presence/status and for attendant console. It is pretty cool but there are scalability issues. Olle E. Johansson wrote: Rich Adamson wrote: So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Not sure, but seems to me it came in about the time Olle and Snom were talking, and Olle was working on sip-2 or something like that. There is support for SUBSCRIBE in the SIP channel. I've been spending time trying to understand this and also asked questions with no answer. My guess is that you can subscribe to an extensions (not sip device) status and get notificiations when the status changes. Would be useful for phones that have line indicators. There's some authorization for doing this, but no ACL list - who can subscribe to what? Sorry for an unclear explanation, I haven't been able to test with any device. Anyone that knows any device that supports SUBCRIBE this way? BTW, isn't SUBSCRIBE also used for SIMPLE, instant messaging? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT:Linux(or *BSD) SNMP tools (Was: Re: rtp sound quality?)
Chris Craft wrote: On Saturday 31 January 2004 21:31, you wrote: <<>> I am just a low level c hack. Before I go out and write any thing to do this snmp admin stuff, are there any linux tools I could use to do this? Net-SNMP (http://freshmeat.net/projects/net-snmp/ , formerly UCB-SNMP or something) is very handy for this. Perfect. Thank you very much. bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PCI expansion slots.
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk > Sent: Saturday, January 31, 2004 10:05 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] PCI expansion slots. > > > > Hello, > > Did anyone use PCI expansion slots such as: > > http://www.cyberresearch.com/store/product/311.2.htm > > I want to know how well does it work with Asterisk FXO/FXS > cards? Also, does FXO/FXS drivers work automatically > (meaning seemlessly recognize the expansion slots) without > any Power/Bandwidth/Interrupt issues? > > Any alternative or information about working (or not working) > baords would be highly appreciated. That's not a "PCI Expansion Slot". It's a passive backplane, designed to host a single-board "industrial" type machine. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?
Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best, Michael Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
[Asterisk-Users] short ringing
Hello.
I'm a bit puzzled at the moment. I have a x100P and TDM400 with 4 modules
(extensions). Asterisk CVS-02/01/04-06:55:30
Part of my extensions.conf says this:
; Zap Phone #1
;
exten => 204,1,Dial(Zap/2,20) ; Ring for 20 seconds
exten => 204,2,Voicemail(u${EXTEN})
exten => 204,3,Hangup ; Unavail voicemail if extension
doesn't answer
exten => 204,102,Voicemail(b${EXTEN}) ; Busy Voicemail if extension is busy
exten => 204,103,Hangup
;
;
; Zap Phone #2
;
exten => 120,1,Dial(Zap/3,20) ; Ring for 20 seconds
exten => 120,2,Voicemail(u${EXTEN})
exten => 120,3,Hangup ; Unavail voicemail if extension
doesn't answer
exten => 120,102,Voicemail(b${EXTEN}) ; Busy Voicemail if extension is busy
exten => 120,103,Hangup
If I call from ext. 120 to ext. 204, it rings for 15 seconds. I call from
ext. 204 to ext. 120, it rings for 10 seconds. In both cases, it announces
the unavail and goes to voicemail.
Why different rings times and why not 20 seconds ?
Regards...Martin
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RE: [Asterisk-Users] TE410P E1 PRI problem
Thanks, the problem was with crc. It is ok now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Maj Sent: Sunday, February 01, 2004 12:31 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P E1 PRI problem On Sat, 31 Jan 2004, Tomica Crnek waxed: > Hi everyone! > > Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a problem with PRI synchronization or PRI to zap communication. > > The card I am using is TE410P, the first port is the one that I use. > > /etc/zaptel.conf > - > # port 1: trunk to telecom > span=1,0,0,ccs,hdb3 Try this instead: span=1,1,0,ccs,hdb3 (Note the second 1, that sets your timing to the telecom.) --Chris -- Chris Maj Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMDI on *
anyone seen or used * voicemail on a PBX system that needs to talk SMDI to the VM host? Dave Packham Dave Packham University of Utah Netcom Campus R&D c. [EMAIL PROTECTED] w. [EMAIL PROTECTED] [EMAIL PROTECTED] Trillian/ICQ#:45818442 MSN [EMAIL PROTECTED] Our Groups Website http://rd.it.utah.edu ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
> >>>What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> >>>
> >>>Been trying stuff similar to:
> >>>exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
> >>>where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
> >>>even try the IP.
> >>>
> >>>Rich
> >>>
> >>
> >>from my extensions.conf:
> >>
> >>;**
> >>[trunk-local]
> >>;**
> >>exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> >
> >
> > The above does not seem to work either. Since the mediatrix has four pstn
> > ports, there must be a way to construct a Dial command that would embed
> > a userid:password, port alias name, or something like that. Just can't find
> > any reference to what that syntax would look like. (The gateway is properly
> > handling incoming pstn calls, just not the outgoing pstn attempts.)
> >
> > Really need to the sip dial command to include...
> > - the string of digits to be called
> > - either a userid:password, or, port alias name (or both)
> > - ip address of the gateway
> >
> > Anybody have a clue what that dial sip syntax would look like
> Yes, it's
> SIP/[EMAIL PROTECTED]
> There's no 'sub-extension'.
>
> So SIP/[EMAIL PROTECTED] is the proper way to go, where extension is
> the string of digits to be called. If the box acts as a SIP proxy, you
> might need to register with a register=> in sip.conf beforehand.
>
> This is like calling any FWD extension. First, register, then place
> a call with
>DIAL(SIP/[EMAIL PROTECTED])
>
> Any pointer to the manual?
No, the manual is very verbose but no * examples at all. The box sells as
either a 323 or sip, with different images (sort of like C7960's) and
different manuals.
The box does not support the "register" function in either direction. I just
tried the * sip register, and got a "501 Not Implemented" with sniffer.
>From what I can tell (box is about 48 hrs old for me), it seems to be a
rather incomplete or just-bare-sip-minimum functionality. It also appears
as though all four ports are treated as a group-of-lines, and one doesn't
have any choice (from a sip perspective) on which port to use for outgoing
calls. Since this one is set up with 1:home, 2:business, 3:outgoing calls
I really need to be able to select which port * is going to use, particularly
since outgoing 'home' long distance calls must use a different port then for
outgoing 'business' calls.
The entire box (4 ports) has only a single IP, so if the dial sip command
doesn't have any additional parameters/strings to destinguish selected ports,
guess I'll return it to the reseller. There appears to be a way to set certain
types of filters on a per port basis in the box, but I can't see how that
could be used to differentiate home vs business calls, etc.
Since I don't know anything about 323, does that control protocol allow some
sort of sub-selection where each port would be addressable? If not, it certainly
seems as though Mediatrix needs to go back to work on their code or something.
Can you think of any other way that * might interact with this thing via sip?
Rich
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Re: [Asterisk-Users] Dial via sip gateway?
Rich Adamson wrote:
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
The above does not seem to work either. Since the mediatrix has four pstn
ports, there must be a way to construct a Dial command that would embed
a userid:password, port alias name, or something like that. Just can't find
any reference to what that syntax would look like. (The gateway is properly
handling incoming pstn calls, just not the outgoing pstn attempts.)
Really need to the sip dial command to include...
- the string of digits to be called
- either a userid:password, or, port alias name (or both)
- ip address of the gateway
Anybody have a clue what that dial sip syntax would look like
Yes, it's
SIP/[EMAIL PROTECTED]
There's no 'sub-extension'.
So SIP/[EMAIL PROTECTED] is the proper way to go, where extension is
the string of digits to be called. If the box acts as a SIP proxy, you
might need to register with a register=> in sip.conf beforehand.
This is like calling any FWD extension. First, register, then place
a call with
DIAL(SIP/[EMAIL PROTECTED])
Any pointer to the manual?
/O
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Re: [Asterisk-Users] Dial via sip gateway?
> >What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> >
> >Been trying stuff similar to:
> > exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
> >where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
> >even try the IP.
> >
> >Rich
> >
> from my extensions.conf:
>
> ;**
> [trunk-local]
> ;**
> exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
The above does not seem to work either. Since the mediatrix has four pstn
ports, there must be a way to construct a Dial command that would embed
a userid:password, port alias name, or something like that. Just can't find
any reference to what that syntax would look like. (The gateway is properly
handling incoming pstn calls, just not the outgoing pstn attempts.)
Really need to the sip dial command to include...
- the string of digits to be called
- either a userid:password, or, port alias name (or both)
- ip address of the gateway
Anybody have a clue what that dial sip syntax would look like
Rich
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Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
Florian Overkamp wrote: Hi, -Original Message- So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Hmm, dunno. Could it be used to have your MSN contacts show as available or no ? :-) After a bit more experimenting, I found that my context for the user included a match-all, so everything was ok. If I tried another context, it answers OK. For my installation, I need a separate context per peer for SIP users if I'm going to support SUBSCRIBE. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
Mike,
I'm hoping one can specify a particular mediatrix "port" in the Dial Sip
command, but haven't found any Dial syntax that would allow passing a
userid/password to the gateway. Since the 1204 provides a AuthUsrPwd on
a per port basis, my guess would be that we either have to pass the Alias
defined for that port or the AuthUsrPwd in the Dial command.
When I attempt
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I get an immediate "407 Proxy Authentication Required" back. However, with
a packet sniffer running, * isn't even sending a packet to the mediatrix.
I'd have to guess and assume * is doing this because the mediatrix isn't
'registered' with *, but the mediatrix was not designed to register anyway.
I'm stuck in the Dial syntax, and can't seem to find any google reference
as to how to pass the needed parameters.
Rich
> Bob, I have a question into mediatrix for this, but maybe you have
> figured it out. I am trying to map a SIP user to a specific PSTN line. I
> have my extensions.conf file as you show below, but on the 1204, it just
> grabs whatever line is available, whereas I want extension 101 to always
> be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
> NetToPstnSourceFilter MIB per port, and their docs hint at using this,
> but the example in the docs describes their FXS to FXO, so I am not sure
> what I would put in that MIB. CallerID info? * calling sip extension
> number? Have you been able to make this work?
>
> On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
> > Rich Adamson wrote:
> >
> > >I'm having a brain fart
> > >
> > >What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> > >
> > >Been trying stuff similar to:
> > > exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
> > >where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
> > >even try the IP.
> > >
> > >Rich
> > >
> > from my extensions.conf:
> >
> > ;**
> > [trunk-local]
> > ;**
> > exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> > exten => _9NXX,2,Congestion
> >
> > [trunk-toll]
> > exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> > exten => _91NXXNXX,2,Congestion
> --
>
>
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Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
Florian Overkamp wrote: Hi, -Original Message- So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Hmm, dunno. Could it be used to have your MSN contacts show as available or no ? :-) Of course! I configured Windows Messenger to work with my Chan_sip2. It's important for messenger that the REALM (configurable in chan_sip2) is the same as the proxy's name. When I add a contact, messenger subscribes to it. And Asterisk answers that anything is online. Florian is online in my server without any peer "florian" and any (at least known by me) extension named "florian". If I add my FWD account, asterisk accepts subscription on it and merrily answers "online" without checking. A positive attitude, indeed. Obviously there's a bug here. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
Hi, > -Original Message- > So, what hardware or use is the SUBSCRIBE method used for in > chan_sip.c? I asked this question a while ago, and got > resounding silence. Maybe someone who is better at > de-tangling C code than I am could take a peek. Hmm, dunno. Could it be used to have your MSN contacts show as available or no ? :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
Rich Adamson wrote: So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Not sure, but seems to me it came in about the time Olle and Snom were talking, and Olle was working on sip-2 or something like that. There is support for SUBSCRIBE in the SIP channel. I've been spending time trying to understand this and also asked questions with no answer. My guess is that you can subscribe to an extensions (not sip device) status and get notificiations when the status changes. Would be useful for phones that have line indicators. There's some authorization for doing this, but no ACL list - who can subscribe to what? Sorry for an unclear explanation, I haven't been able to test with any device. Anyone that knows any device that supports SUBCRIBE this way? BTW, isn't SUBSCRIBE also used for SIMPLE, instant messaging? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
