[Asterisk-Users] Conference calls on Pingtel Phones
Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk hardphone
I am using Pingtel phones right now and find that they work well and have some cool features. Got most everything working except conferencing and we're working on that. I love some of the ringtones (i.e. 007 theme, Beavis & Butthead, Homer Simpson Doh! etc) Sincerely, Andy Hester Consero -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of ChrisSent: Wednesday, June 25, 2003 2:45 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk hardphone I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented Asterisk with a hardphone? Which one?
[Asterisk-Users] Help! Problems talking to upstream switch
Hi, Please let me know if you have any ideas - I am taking wild guesses now Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP phones save a few fax machines. The customer has 7 digit unverified account codes on the trunks for billing purposes. The Problem: As I watch the console, I see calls coming in for exten "73" or "708" or "08" or "730" although most come in correctly (ie "7308"). My carrier has verified numerous times that they are sending 4 digits. I have 40 DID numbers that need to be routed and they are all in the 73xx range. I need to know anything that would cause my box randomly not to hear all 4 digits on occasion. Also, I have had trouble with people who dial out getting a congestion signal mainly on Long Distance numbers. The person would dial the number 4 or 5 times and get congestion then it might go through. Both of these conditions seem to be happening only about 10-20% of the time. What I have done: Moved a T100P card to its own IRQ to prevent problems with interrupts - Did not solve either issue. On the second try, got the carrier to change the way their switch chooses channels for incoming calls to prevent "glare" - This MAY have fixed the outgoing long distance issue as it seems to have gone away( although it doesn't seem logical to me that this would affect only LD) but did not fix incoming calls. Has anyone else had problems getting all of the digits that the Telco sends? Thanks, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help! Problems talking to upstream switch
Thanks for the info... I've answered your questions below. I am not experienced with telecom at this level (yet), but this sounds like really good info to quiz XO's switch tech over. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Sunday, June 29, 2003 9:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help! Problems talking to upstream switch Whats the sound quality like on the calls especially when multiple calls are going? No problems with sound quality save the slight echo on calls over the TDM circuit. On my home system, I had a problem that DTMF was sometimes not correctly recognized and would either dial incorrectly or not at all. It was evedent when dial tone was played that it crackled. Also when multiple calls where running each would start to sound like crap. This was later tracked down to a timing problem. Since you mention using a CAC with 2 ports on it and a router, I'm going to assume you have a ADIT 600. Make sure the Adit is set to take timing from the telco, and then make sure you are set to take your timing from it. Check to make sure each of the T1 ports a:1 and a:2 are set to take timing from a:1 if it is your telco port. This will keep you from slipping and causing potential problems. I haven't looked around inside the CAC yet since it is XO's, not sure if it is an ADIT 600 but it sounds like the same unit. I am set to "0" for timing and "0" for LBO in Zaptel.conf I assume this is correct for * and that I need to verify the a:1/a:2 timing settings in the CAC unit? Come to think of it, there is a way to test this without bringing the T1s down. The Adit 600 has a show performance command, I may be wrong, but I'm sure it was performance, anyways it allows you to see slips and bipolar violations and a couple other stats. This was beneficial for me as the T100P didn't report problems but the Adit did. Can you give me a brief idea of what slips and bipolar violations are? Home this helps. I am glad to find that someone knows more than my little knowledge of the subject! To here the techs talk you'd think that they'd never run into anything like this before. On Sun, 2003-06-29 at 20:59, Andy Hester wrote: > Hi, > Please let me know if you have any ideas - I am taking wild guesses now > Here is the situation: > > I put in Asterisk for a local customer. I have Fractional T-1 with 12 > Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a > chanel bank although it only has 2 outputs) in a CAC unit. The unit also > has a router card that runs the data side. My extensions are all SIP phones > save a few fax machines. The customer has 7 digit unverified account codes > on the trunks for billing purposes. > > The Problem: > > As I watch the console, I see calls coming in for exten "73" or "708" or > "08" or "730" although most come in correctly (ie "7308"). My carrier has > verified numerous times that they are sending 4 digits. I have 40 DID > numbers that need to be routed and they are all in the 73xx range. I need > to know anything that would cause my box randomly not to hear all 4 digits > on occasion. Also, I have had trouble with people who dial out getting a > congestion signal mainly on Long Distance numbers. The person would dial > the number 4 or 5 times and get congestion then it might go through. Both > of these conditions seem to be happening only about 10-20% of the time. > > What I have done: > > Moved a T100P card to its own IRQ to prevent problems with interrupts - Did > not solve either issue. > > On the second try, got the carrier to change the way their switch chooses > channels for incoming calls to prevent "glare" - This MAY have fixed the > outgoing long distance issue as it seems to have gone away( although it > doesn't seem logical to me that this would affect only LD) but did not fix > incoming calls. > > Has anyone else had problems getting all of the digits that the Telco sends? > > Thanks, > Andy Hester > Consero > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Help! Problems talking to upstream switch
Title: [Asterisk-Users] Help! Problems talking to upstream switch This helps alot! I believe that it is an ADIT 600, and I definitely want to adjust those gain settings. I'll ask Martin about the timeout on DNIS, although it would seem that the fact that I have observed the loss of only the 2nd tone, for example, leads me to believe that it is not the timing out. Thanks, Andy Hester Consero -Original Message-From: Tim McQueen [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Sunday, June 29, 2003 10:11 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Help! Problems talking to upstream switch I'm new to *, but I've dealt with this issue on other switches. It sounds like either you are timing out getting DNIS information from your CO, or you are having trouble hearing the DTMF tones that are pulsed to you during the call setup process. Someone else on the list may know: is it possible to 1) configure the timeout for waiting on DNIS and 2) is it possible to change the Rx gain on the TDM cards? It's my understanding that the circuit is going throught the channel bank, which is acting like a drop-and-insert CSU by forwarding the 12 channels with voice to your * box. You mentioned that this was a Carrier Access channel bank, is it an ADIT 600? There are send and recieve gain settings on the channel bank unit that you might want to play with. HTH, HAND -Tim -Original Message----- From: Andy Hester Sent: Sun 6/29/2003 8:59 PM To: [EMAIL PROTECTED] Cc: Subject: [Asterisk-Users] Help! Problems talking to upstream switch Hi, Please let me know if you have any ideas - I am taking wild guesses nowHere is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it achanel bank although it only has 2 outputs) in a CAC unit. The unit alsohas a router card that runs the data side. My extensions are all SIP phonessave a few fax machines. The customer has 7 digit unverified account codeson the trunks for billing purposes.The Problem: As I watch the console, I see calls coming in for exten "73" or "708" or"08" or "730" although most come in correctly (ie "7308"). My carrier hasverified numerous times that they are sending 4 digits. I have 40 DIDnumbers that need to be routed and they are all in the 73xx range. I needto know anything that would cause my box randomly not to hear all 4 digitson occasion. Also, I have had trouble with people who dial out getting acongestion signal mainly on Long Distance numbers. The person would dialthe number 4 or 5 times and get congestion then it might go through. Bothof these conditions seem to be happening only about 10-20% of the time.What I have done:Moved a T100P card to its own IRQ to prevent problems with interrupts - Didnot solve either issue.On the second try, got the carrier to change the way their switch chooseschannels for incoming calls to prevent "glare" - This MAY have fixed theoutgoing long distance issue as it seems to have gone away( although itdoesn't seem logical to me that this would affect only LD) but did not fixincoming calls.Has anyone else had problems getting all of the digits that the Telco sends?Thanks,Andy HesterConsero___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users <>
RE: [Asterisk-Users] Newbie, Loaded Asterisk can't figure out manual
John, I agree that the manual in its current state is not particularly comprehensive or cohesive shall we say. Here are my suggestions: 1. Configure Zaptel.conf in /etc 2. Configure Zapata.conf in /etc/asterisk to match your definitions in zaptel 3. Configure your end user devices in their config file as appropriate. 4. Write extensions.conf and voicemail.conf Should be ready to test. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Your Name Sent: Sunday, June 29, 2003 7:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie, Loaded Asterisk can't figure out manual Just loaded it yesterday running on TDM400P and X100P. I have also loaded the sample setttings. 1. What's the first thing you guys do? Change .conf files or do it from CLI? 2. Just trying to get it up and running to see if everything works. Do I setup Extensions first? 3. Can I just remove some of the ; within the .conf files to get a simple PBX working? I would appreciate your thoughts and knowledge! John ___ 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] Help! Problems talking to upstream switch
Steven, I thought that "1" would mean that my T100P card would set the timing for the line. Is this incorrect? If I am reading this wrong then please set me straight. My carrier has their end set to be the sync source. If I set the timing to "1", won't that conflict? My line is set up esf/b8zs, so does that mean I can ignore all bipolar violations, or just that a certain number are to be expected? Also, shouldn't the switch tech from my carrier be knowledgeable about these things and trying to help me match up to their settings? I appreciate you indulging me this evening. I'm in somewhat of a bad spot with my customer for a variety of reasons, most beyond my control and I am trying to get their problems resolved asap. Thanks again. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Monday, June 30, 2003 12:04 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Help! Problems talking to upstream switch > > > Use 1 for timing. You actually do have a timing source to sync to. > > As for slips and bipolar violations... > T1s are just high speed serial lines. A sleep is when you loose sync > with the far side and when you see a 1 come across the line, you may not > know which bit it was for. This would be a slip. Bipolar violations are > a part of the signaling, but can also be errors. A T1 alternates the > polarity of the 1 pulse to allow the line to run farther on lower > voltage. Just doing alternating polarity is AMI or Alternate Mark > Inversion. A bipolar violation is when a bit is received as the same > polarity as the last bit received. On an AMI line a bipolar violation is > an error. On a B8ZS, bipolar violations are intentionally inserted into > the line to keep the line from transmitting too many 0's in a row and > contributing to a slip. When set for B8ZS the "error" is somewhat > expected and ignored. > > On Sun, 2003-06-29 at 23:19, Andy Hester wrote: > > Thanks for the info... I've answered your questions below. I am not > > experienced with telecom at this level (yet), but this sounds > like really > > good info to quiz XO's switch tech over. > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Steven > > Critchfield > > Sent: Sunday, June 29, 2003 9:40 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Help! Problems talking to upstream switch > > > > > > Whats the sound quality like on the calls especially when multiple calls > > are going? > > > > No problems with sound quality save the slight echo on calls > over the TDM > > circuit. > > > > > > On my home system, I had a problem that DTMF was sometimes not correctly > > recognized and would either dial incorrectly or not at all. It was > > evedent when dial tone was played that it crackled. Also when multiple > > calls where running each would start to sound like crap. This was later > > tracked down to a timing problem. > > > > Since you mention using a CAC with 2 ports on it and a router, I'm going > > to assume you have a ADIT 600. Make sure the Adit is set to take timing > > from the telco, and then make sure you are set to take your timing from > > it. Check to make sure each of the T1 ports a:1 and a:2 are set to take > > timing from a:1 if it is your telco port. This will keep you from > > slipping and causing potential problems. > > > > I haven't looked around inside the CAC yet since it is XO's, > not sure if it > > is an > > ADIT 600 but it sounds like the same unit. I am set to "0" for > timing and > > "0" for > > LBO in Zaptel.conf I assume this is correct for * and that I > need to verify > > the a:1/a:2 timing settings in the CAC unit? > > > > Come to think of it, there is a way to test this without bringing the > > T1s down. The Adit 600 has a show performance command, I may be wrong, > > but I'm sure it was performance, anyways it allows you to see slips and > > bipolar violations and a couple other stats. This was beneficial for me > > as the T100P didn't report problems but the Adit did. > > > > Can you give me a brief idea of what slips and bipolar violations are? > > > > Home this helps. > > > > I am glad to find that someone knows more than my little > knowledge of the > > subject! > > To here the techs talk you'd think that they'd never run into > anything like > > this before. >
[Asterisk-Users] That is not a valid conference number meesage
I've just started trying to use this functionality and I get the invalid conference number message. Any ideas? I started out with: exten => 7315,1,Meetme,1234 and confno = 1234 and then tried: exten => 7315,1,Meetme and confno = 1234 and enter 1234 at prompt. All give the same message. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Coding Time/Date announce on voicemail
I have a customer asking to have the voicemail give the time and date a message is received when the message is played. Anyone have an idea of how big of a project it will be to code this into the voicemail app? Any suggestions would be greatly appreciated. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank configuration
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Marty Mastera > Sent: Thursday, July 10, 2003 6:34 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Channel Bank configuration > > > Hello, > > I don't have any experience with channel banks and would appreciate any > feedback on my theory outlined below: > > We have a single T1 entering the building with channels 1-12 being voice > lines and 13-24 being a 768k internet connection. This T1 terminates to > an Adit 600 (T1-1). > > Here's what I know. Channels 11-12 go out the Adit 600's 25-pair > connector to a wiring block (and eventually to 2 fax machines - I assume > this is to have the fax machines bypass the currently installed phone > switch). The data comes out the Router card on the Adit and into > our network. > > The currently installed phone system is an NEC NEAX2000 IVS box which is > connected by CAT5 to the Adit 600's T1-2 port. (I am assuming that voice > channels 1-10 are mapped to the Adit T1-2 and getting to the NEC this > wayThere is also a 25 pair cable leaving the NEC and terminating on a > wiring block for the desk phones. > > So my assumption is that channels 1-10 are mapped as FXO onto the Adit's > 2nd T1 port, channels 11-12 to the Adit's 25-pair connector and 13-24 to > the router card for data. > > This leaves the NEC box to handle the FXS (and hence why it is directly > connected to the phones). So far so good... > > When I replace the NEC box with an * box/T100P, I'm thinking that I will > have to map Channels 1-12 to the T1-2 port and map the 8 Adit FXS > channels on the T1-2 port to the Adit's 25-pair for the Adit's FXS > capabilitythen run that T1 into the T100P and configure * to route > between the FXO and FXS channels appropriately. yes - map channels 1-12 to T1-2 on the Adit, but what are the "8 Adit FXS channels on the T1-2 port"? Also, what type of desk sets are you planning to use with *? > > Does this sound right? I'm trying to understand if the channel bank uses > the T1 from the Adit for both FXO and FXS channels! > > Thanks, > > Marty > > > > > -- > Jumping through hoops to get E-mail on the road? > You've got two choices: Join the circus, or use Molly Mail. > > Molly Mail -- http://www.mollymail.com > -- > Having trouble sending email from different locations ? > Need a single outgoing mail server that will work from anywhere ? > > Set it to smtp.com and never have to change it again ! > > http://www.smtp.com > -- > > > ___ > 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] Channel Bank configuration
Marty, You will not be able to use these phones with *. As far as I know, there are no digital desksets that work with asterisk. This leaves you with either analog sets or VoIP sets. If you want to use analog sets, the config that you mentioned would probably work, but you would be dependant on your phone co. to make any changes or help with troubleshooting. This doesn't sound too good to me but then again, it might be less expensive. If you want to go VoIP, you could get the VoIP desk sets and plugem into an ethernet switch along with your * box. Whether or not you split out the analog lines at the ADIT or at the * box really just depends on whether you want to do any management on those calls. If you do want to run those through asterisk, you could get one of their new tdm cards instead of sending those channels back to the ADIT. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Marty Mastera > Sent: Thursday, July 10, 2003 11:25 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Channel Bank configuration > > > Hello Andy, thank you very much for your response... > > The "8 Adit FXS channels on T1-2" that I was referring to is the 8 port > FXS card that is installed in the Adit. I was trying to describe our > voice channels 1-12 entering the * box via T1-2 and having * route up to > 6 extensions back via T1-2 to the Adit's 8 port FXS card and ultimately > to the phones via the 25-pair connector. > > Is this legitimate? > > I may have a problem with the phonesThe current phones are an NEC > model (DtermE) which say "for NEC pbx only" on the backdoes anyone > know if they will work? Otherwise can a decent business phone be > recommended (with transfer, hold, callerid, etc...capabilties)? > > Thanks again Andy, > > Marty > > On Thu, 10 Jul 2003 23:02:22 -0500 "Andy Hester" wrote: > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] Behalf Of > Marty Mastera > > > Sent: Thursday, July 10, 2003 6:34 PM > > > To: [EMAIL PROTECTED] > > > Subject: [Asterisk-Users] Channel Bank configuration > > > > > > > > > Hello, > > > > > > I don't have any experience with channel banks and would > appreciate any > > > feedback on my theory outlined below: > > > > > > We have a single T1 entering the building with channels 1-12 > being voice > > > lines and 13-24 being a 768k internet connection. This T1 > terminates to > > > an Adit 600 (T1-1). > > > > > > Here's what I know. Channels 11-12 go out the Adit 600's 25-pair > > > connector to a wiring block (and eventually to 2 fax machines > - I assume > > > this is to have the fax machines bypass the currently installed phone > > > switch). The data comes out the Router card on the Adit and into > > > our network. > > > > > > The currently installed phone system is an NEC NEAX2000 IVS > box which is > > > connected by CAT5 to the Adit 600's T1-2 port. (I am assuming that > > voice > > > channels 1-10 are mapped to the Adit T1-2 and getting to the NEC this > > > wayThere is also a 25 pair cable leaving the NEC and > > terminating on a > > > wiring block for the desk phones. > > > > > > So my assumption is that channels 1-10 are mapped as FXO onto > the Adit's > > > 2nd T1 port, channels 11-12 to the Adit's 25-pair connector > and 13-24 to > > > the router card for data. > > > > > > This leaves the NEC box to handle the FXS (and hence why it > is directly > > > connected to the phones). > > > > So far so good... > > > > > > > > When I replace the NEC box with an * box/T100P, I'm thinking > that I will > > > have to map Channels 1-12 to the T1-2 port and map the 8 Adit FXS > > > channels on the T1-2 port to the Adit's 25-pair for the Adit's FXS > > > capabilitythen run that T1 into the T100P and configure * to route > > > between the FXO and FXS channels appropriately. > > > > yes - map channels 1-12 to T1-2 on the Adit, but what are the > "8 Adit FXS > > channels on the T1-2 port"? > > > > Also, what type of desk sets are you planning to use with *? > > > > > > > > > > Does this sound right? I'm trying to understand if the c
RE: [Asterisk-Users] New Member
I am using Redhat 9.0 and have no problems...Redhat is a good choice if you want to get going fast. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Erik Anderson > Sent: Saturday, July 12, 2003 10:31 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] New Member > > > Redhat 8 > > We are having problems with RH 9. It is too bleeding edge. > > Erik > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of jltaylor > > Sent: Saturday, July 12, 2003 10:00 AM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] New Member > > > > > > Greetings, > > I'm ready to start and setup Asterisk. > > > > Any preference on which Linux to use? > > Windows & FreeBSD in use here. > > > > I'd like to get up and running as quickly and easily as possible. > > > > Thanks > > > > -- > > James Taylor > > [EMAIL PROTECTED] > > 903-793-1953 > > > > -- > > ___ > > 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] time and date stamp in voicemail
I have searched and not located this patch...is there a specific place that I need to look, or a specific file name? Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of John Todd > Sent: Thursday, July 24, 2003 6:14 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > >Hi, > > > >I see that there's been some very light discussion on having a > standard time > >and date stamp in VM. How can I implement it today? (About to offer a > >system to a customer but they need the stamp to tell when people called.) > > > >Thanks, > >-- > > > >Steve > >__ > >This sig is pending approval > > Tilghman Lesher had a well-written patch he posted to the list a few > weeks ago for Voicemail, which I've been using without difficulty. > He has said he's going to work on Voicemail2, so I am hoping to see > that soon, and then integration into the main CVS tree after some > testing. > > Not only does the patch handle generic time and date stamps, but it > allows customizable timezones, announcement strings, and uses > standard UNIX-ish time code macros. Very slick, and really necessary > for voicemail systems that happen to have users in multiple timezones. > > If you are looking for words to match Tilghman's patch, see the > phrases I have submitted (and donated by a generous grant by > VoicePulse) as public domain, recorded by Allison Smith: > http://www.loligo.com/asterisk/sounds/ > > JT > ___ > 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] time and date stamp in voicemail
Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Dan > Sent: Friday, July 25, 2003 2:52 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > Hi, > > This page does not exist... > > Thanks, > Dan > > - Original Message - > From: "Steven Critchfield" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, July 25, 2003 8:38 AM > Subject: RE: [Asterisk-Users] time and date stamp in voicemail > > > > Try looking drunkencoder.com/asterisk > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Tilghman, Thanks alot for posting that. I'll check it out Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tilghman > Lesher > Sent: Friday, July 25, 2003 10:48 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > On Friday 25 July 2003 14:12, Andy Hester wrote: > > Dan, > > the page is actually http://asterisk.drunkcoder.com/patches/ . > > However, I didn't see the patch there. > > I just added it. It's available there now. Note that there are three > files: a patch, sounds, and some instructions. > > -Tilghman > > ___ > 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] time and date stamp in voicemail
Tilghman, I applied your voicemail_prompts patch and it works like a charm. Thanks for donating the code and thanks to those that donated the voice prompts! Another win for Asterisk Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tilghman > Lesher > Sent: Friday, July 25, 2003 10:48 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > On Friday 25 July 2003 14:12, Andy Hester wrote: > > Dan, > > the page is actually http://asterisk.drunkcoder.com/patches/ . > > However, I didn't see the patch there. > > I just added it. It's available there now. Note that there are three > files: a patch, sounds, and some instructions. > > -Tilghman > > ___ > 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] time and date stamp in voicemail
Tilghman, I'm not sure how to use this logic. Would this be for something like, for example, deleting of forwarding a message that a certain age? Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Tilghman > Lesher > Sent: Sunday, July 27, 2003 11:24 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] time and date stamp in voicemail > > > On Saturday 26 July 2003 21:06, Andy Hester wrote: > > Tilghman, > > I applied your voicemail_prompts patch and it works like a charm. > > Thanks for donating the code and thanks to those that donated the > > voice prompts! Another win for Asterisk > > Is anybody at all using the variable substitution and/or the expression > logic at all? I'd like to know if and how well it works for you. > > -Tilghman > > ___ > 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] time and date stamp in voicemail
Tilghman,
I will implement this... I think its a fairly important feature. Let me
know what you find in your testing.
Sincerely,
Andy Hester
Consero
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Tilghman
> Lesher
> Sent: Monday, July 28, 2003 1:24 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] time and date stamp in voicemail
>
>
> On Sunday 27 July 2003 23:57, Andy Hester wrote:
> > Tilghman,
> > I'm not sure how to use this logic. Would this be for something
> > like, for example, deleting of forwarding a message that a certain
> > age?
>
> No, this would be used in something like:
>
> central=US/Central|'vm-received' $[${DIFF_DAY} < 7]?A:BdY
>
> where it would read only the day of the week if the message were less
> than 7 days old and otherwise read the month, day, and year.
>
> Actually, now that I'm looking at the code, I don't think this works,
> so I'm going to do a bit more work and test it myself.
>
> -Tilghman
>
> ___
> 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] Offering an Asterisk Documentation and FAQ Portal
I have been planning to suggest this as well, but I would recommend setting up a zope site... If you set it up in zope you can have alot of collaboration very easily. You could, for instance, designate certain people who have expertise in a certain config/technology as project coordinator for that area of documentation. As well you could, as with any other system, create areas for specific technologies etc for the purpose of being easy to navigate. And of course the main benefit is the ability to have as many people collaborate as you wish, and still maintain organization of data with very low administration. Have a look...I think it would make all of our lives much easier. www.zope.org example site http://www.openparadigms.com allows you to create an account and post pages etc... Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer > Sent: Monday, July 28, 2003 4:09 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Offering an Asterisk Documentation and FAQ > Portal > > > Agreed. We're more than happy to host it. The problem is the writing of > it :) > > Mark > > On Mon, 28 Jul 2003, Scott Stingel wrote: > > > Just a suggestion, but wouldn't it be more appropriate for > Digium to host > > the documentation? > > > > I think the missing link here is someone who will write (and > illustrate) the > > documentation. All of this open source software is great > because it's free > > - but commercial users and others would certainly appreciate > the time saved > > by referring to a nice doc set. I for one would have been > willing to pay > > for a reasonable documentation set, especially at the outset. > > > > Maybe this is a commercial opportunity for a good tech writer - maybe > > working in collaboration with Digium. > > > > ...just my 2 cents! > > > > Scott > > > > > > > > Scott M. Stingel > > Emerging Voice Technology Inc. > > Palo Alto, California and London, England > > > > Email: [EMAIL PROTECTED] > > URL:www.evtmedia.com <http://www.evtmedia.com/> > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Damian Flynn > > Sent: Monday, July 28, 2003 8:17 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Offering an Asterisk Documentation > and FAQ Portal > > > > > > Hi, > > > > I have resources available to host a portal specifically for > the Asterisk > > system, to help correlate documentation, FAQ's and How To > > > > I am new to Asterisk, and my hardest work is in locating information on > > using or configuring the software. > > > > Would Mark, John or any of you feel this would be of benefit to host? > > > > I am offering a PHP-NUKE portal for this, (Unless you know of a better > > solution!) > > > > Regards > > Damian > > > > > > ___ > 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] RE Pingtel Phones
Hello, Is anybody else out there using pingtel phones? If so, I like to hear your experiences... Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Analog sets for use w/*
Hi All, I am considering testing out some analog sets with * for a customer and thought I would ask what analog phones are in use? The customer would require the usual business functionality ie hold, conference calling, and preferably a soft key to vm and line apearances(correct terminology?) in order for secretary to see if their boss is on the phone before transferring a call, etc. I would appreciate haering any of your experiences. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Analog sets for use w/*
TC, Have you used these phones? They seem to be pretty nifty... I'm wading through the documentation to see how well they would integrate. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of TC > Sent: Thursday, July 31, 2003 8:11 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Best Analog sets for use w/* > you might want to take a look at http://smartalk.ca/nrgover.htm > these are analog phones designed for pc-pbx's, that have a rj-45 cable > pins 1-8 are looped as a system bus to allow you to intercom, & see which > sets are in use... the reception phone SE310 has more soft keys & larger > display > they all have the ability to program the soft keys > > > ___ > 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] list proposal
Perhaps there is another way to cut down on increased traffic... Specifically, I would go back to the suggestion of a collaborative website for documentation. Collecting info and organizing into Howto's would reduce the number of times people ask the same questions. Also, the documentation could grow as quickly as the project. Unfortunately, I don't have a place to host it currently. Ideally, the list would just be for issues that aren't already addressed. Any one else interested in this? Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Friday, August 08, 2003 1:25 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] list proposal > > > With the increased traffic as of late, I'm wondering if it is time to > split the list again. Specifically I am wondering if it should be split > along the various VoIP protocols and zap hardware, then leave a general > list that does configuration other than VoIP related? > > The hope is that those asking SIP or H323 questions could get help from > the various supporters while the main list can deal with transport > neutral content like extension logic and voicemail configs. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > 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] IP phone recommendation
Nathan, I am using the Pingtel phones at a customer site. I should be able to give a report in a couple of days Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Nathan > Littlepage > Sent: Wednesday, August 13, 2003 8:15 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] IP phone recommendation > > > Has anyone had the opportunity to use a PingTel phone with Asterisk? > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list proposal
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Sunday, August 10, 2003 10:31 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] list proposal > > > On Sun, 2003-08-10 at 21:25, Andy Hester wrote: > > Perhaps there is another way to cut down on increased traffic... > > > > Specifically, I would go back to the suggestion of a > collaborative website > > for documentation. Collecting info and organizing into Howto's > would reduce > > the number of times people ask the same questions. Also, the > documentation > > could grow as quickly as the project. Unfortunately, I don't > have a place > > to host it currently. Ideally, the list would just be for issues that > > aren't already addressed. Any one else interested in this? > > While it still needs to be done, the majority of those type questions > will still happen as the newest users still don't use google until told > to do so. There will always be some of this, and I agree that most people won't search for an answer if they can have someone else hand it to them. However, the idea is to make it such that they don't have to search for the answer among thousands of other posts, but rather just read the instructions. And like you said, it needs to be done anyway, so why not try to make the documentation as complete as possible and maybe post a large warning, don't ask questions of the list if you haven't read the documentation! And place this documentation or a link to it before the info on how to sign up for the mailing lists. I imagine many new comers never even see the google search box down the page. It doesn't really bother me that much when those questions are posted mind you, but I just think it is a really bad idea to use a mailing list as the repository for documentation. I know that documentation exists, but it is so incomplete that someone is always telling documentation seekers to "search the list". We should try to move the documentation away from the list and into a seperate place. As new or unaswered questions come up, the documentation can be easily updated if it is collaborative. Anyway, I don't mind splitting the lists as I will just subscribe to those as well. It might work out well as I had also suggested that the documentation be split out by technology also. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI Phones
I am considering switching my Asterisk implementation from SIP to ADSI. I have the channel bank and a T100P for 24 analog stations. Currently my phones crash often. I have some questions though, because I don't want the users to be disapointed again. 1. What ADSI phone do you use (in production:)), what is required to get them going? 2. Can any of the ADSI phones monitor another extension to display a busy indicator for that extension? 3. Is there any standard business feature that you can't provide with a 1 line ADSI phone + Asterisk(exluding tons of hacking)ie hold,transfer by softkey,voicemail by softkey, conference call, voicemail MWI etc.? Is there any advantage to having a 2 or even 3 line ADSI phone? 4. It seems the receptionist would be very limited in routing calls. Voice menu is probably not an option. How have you dealt with that? Any reason not do something like this: exten => 100,1,Dial,Zap/1|15 exten => 100,2,Voicemail,u100 exten => 100,102,Wait|10 exten => 100,103,Play,wait.gsm(not sure if this is the right syntax, but you get the idea) exten => 100,104,Dial,Zap/1|15 exten => 100,105,Voicemail,u100 exten => 100,205,Voicemail,b100 5. Share a source for buying them? Any tips or info/configs would be greatly appreciated. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Tone length
Please point me in the right direction as to where to increase the length of time I am sending DTMF tones on my T100P. I can't remember where/how I set it last time, and I need to do a fresh install. I'd sure appreciate it :) Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Provisioning CO lines
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Mike Ciholas > Sent: Thursday, August 21, 2003 10:21 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Provisioning CO lines > > > > Hi all, > > This is a NEWBIE question, so all you experienced types that are > tired of stupid questions can move on... > > I've pretty much given up trying to do my entire phone system > over IP (including local service), so I have to select and > provision my local CO lines. I need about 10-12 lines which can > be POTS lines, of course. But, I thought, why not get something > digital and expandable like a DS1, PRI, T1 or whatever they call > it with 23 or 24 channels of 64 kbps voice. It seems like it > would be simpler for me to deal with this (and better quality) > and it *should* be simpler for the phone company, too. > > However, while everyone can sell me POTS lines, when I ask about > getting these in some sort of digital muxed interface, I seem to > confuse the providers. In one case, I was able to get something > called "channelized T1" which cost a lot and did not actually > include the "phone" service for any of the channels, that was > additional. So the cost to go from POTS lines to something > digital was extreme, so much more than I can't understand why > anyone would have T1 voice interfaces, yet all the PBXes have > this and it seems commonly used. I must be doing this "wrong". > > Okay, so I need help with: > > 1. Understanding terminology so I can ask for the "right thing". > > 2. Advice on when it is reasonable to go POTS versus something > else and what that something else is. > > 3. Feedback on what others are doing with 10-12 lines in the US > that may want to expand to ~20 lines. > > 4. Interfacing so many POTS lines to Asterisk. I guess that > means an FXO channel bank to T1 card? Kind of stupid to go > digital/analog/digital in the last 100 feet. > > Help? > > -- > Mike Ciholas(812) 476-2721 voice > CIHOLAS Enterprises (812) 476-2881 fax > 2626 Kotter Ave, Unit D [EMAIL PROTECTED] > Evansville, IN 47715http://www.ciholas.com > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I opted for an "integrated T-1" for 1 customer who needed about 12 lines. I configured it with 12 lines voices and 768k data. Chances are you need this kind of bandwidth if you need 12 phone lines. Combining it on 1 T-1 can make it a little more cost effective and of course one of the big advantages is reliability over dsl. They should be able to provide equipment that will give you 2 T-1 outputs, one of wich you just go straight into a T110P. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Default Flash Time
Anyone know offhand what the default flash time is? Where to find and adjust if necessary? Going to test out some analog sets with * and wanted to know. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PCI X100P card interrupt problems
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Ajit M > Kallingal > Sent: Wednesday, August 27, 2003 1:29 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] PCI X100P card interrupt problems > > > My X100P card seems to have interrupt clashes with my Sound card, > any ideas > to prevent this ? > > Thanks and Regards > Ajit > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users Ajit, Disable any unnecessary devices(ie serial or parallel port or usb) in your bios and put your X100P on its own interrupt. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call parking -- what was the key combination?
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan > Caruana > Sent: Friday, September 05, 2003 9:37 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] call parking -- what was the key > combination? > > > what i'm asking is what is the key sequence > you have to dial for the transfer .. > > it was something like *7# if I remember > well, I know I had it working, but the client > lost the paper I wrote it on for him, and I can't > trace the email I got it from! > > cheers > Dave I think its just # and then dial the number for parking ie #700 Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Creating documentation using a web interface
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen > Sent: Saturday, September 06, 2003 9:42 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] OT: Creating documentation using a web > interface > > > Hello all, > > I would like to document some things I am doing with asterisk, but would > prefer to do this from a web interface. I am unfamiliar with any > software that allows you to create online documentation from a web > interface. Ideally I will be able to create documentation online from a > browser, which then when saved, is immediately ready to be read online. > Perhaps I can setup different authors who are also allowed to create > documentation, or have a section where users to the site can create > their own documentation and submit it for inclusion. A section to > submit documentation edits would be nice, as well as maybe a history > timeline or something like that? > > Just some thoughts. If you know of something like this, please let me > know. In the meantime, I'll be googling some more. > > Thanks, > > -- > Leif Madsen - Telecommunications Technology > Sheridan College - Trafalgar Campus > ICQ: 3445119 > FWD: 18924 > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users Check out Zope. It can be really complex or fairly simple. http://www.zope.org Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any way to get out of a remote console without stopping *
This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any way to get out of a remote console without stopping *
Wow look at the choices :) Thanks everyone for the info. I'll try them out. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Marketing Digium/Asterisk
The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies who would benefit from IP trunking - perhaps by industry segment. ie what type of company is likely to have multiiple locations with lots of interoffice communications. Also, I would love to target Centrex customers. I put in an Asterisk system for a local company that had been on Centrex. In short, it will save them $18,000.00 per year (and did pretty well for myself too:)). Any way to ID centrex customers? I'd be glad to share any tips ideas etc. as well if there is any interest. Just a thought. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Marketing Digium/Asterisk
comments inline > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > [EMAIL PROTECTED] > Sent: Friday, October 10, 2003 11:05 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Marketing Digium/Asterisk > > > You need to remember you are looking at it different to what they would. > > > > Regards Mick > Yes, this is true, however, I didn't mean to imply that I wouldn't aggressively market the benefits to the potential customer, but rather that I wanted to find a way to id companies who have the most to benefit from this type of solution. Thanks for the reminder though,I need it as I am not much of a salesman. Andy > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester > Sent: Saturday, 11 October 2003 1:31 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Marketing Digium/Asterisk > > > The benefits of * are obvious so that part of the marketing an * > solution is easy. Anybody care to share ideas on how to target > companies who would benefit most from */Digium? It seems to me that it > would be an easy sell to small/medium companies who need advanced > features such as ip trunking, IVR, Conference bridging, etc., etc. > > I would like to find a way to identify multi-location companies > who would benefit from IP trunking - perhaps by industry segment. ie > what type of company is likely to have multiiple locations with lots of > interoffice communications. > > Also, I would love to target Centrex customers. I put in an > Asterisk system for a local company that had been on Centrex. In short, > it will save them $18,000.00 per year (and did pretty well for myself > too:)). Any way to ID centrex customers? > > I'd be glad to share any tips ideas etc. as well if there is any > interest. Just a thought. > > Sincerely, > Andy Hester > Consero > (817)375-1244 > (817)937-7977 > > ___ > 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] Marketing Digium/Asterisk
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > [EMAIL PROTECTED] > Sent: Friday, October 10, 2003 11:28 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Marketing Digium/Asterisk > > > > That's OK > > Easy to get involved and lose sight of the reason these customers > > A couple of companies that I found to approach were > > > Companies with multiple offices > > With lots of calls between them and spending thousands of dollars a > month. > > Just food for thought > > > Regards Mick Yes, but what type of business did they do? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] don't use Pingtel -was Is this Hardaware Enough for Asterisk ?
-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Tarun BankaSent: Sunday, October 12, 2003 4:17 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Is this Hardaware Enough for Asterisk ? Hello,We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you.1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa2. Wildcard T100P interface card, that will connect Asterisk server to Our Nortel Switch SL-1003. Wildcard TDM400P that gives us 4FXS ports for 4 Analog Phones4. Server 1.8GHz or more P 4 1GB RAM5. T1 Cable.Please let me know if I am missing anything. Best regards,Tarun Tarun, Do NOT buy Pingtel phones for use with Asterisk. I am pulling my installation this week, and another guy I know has had to roll back his large install as well. These phones worked well with Asterisk in a test environment but later manifested many erratic problems upon use in production (mainly phone lock ups). I have recieved no help from Pingtel to resolve the issues other than to suggest that the problems are with * and suggest purchasing their proprietary soft switch and a seperate gateway. These phones have some cool features but do not expect them to stand behind them unless there have been major changes. You have been warned. Sincerely, Andy
RE: [Asterisk-Users] I give up!!
Dave, I sympathize greatly with your plight. I just had a similar experience myself, although I managed to salvage it before I was asked to remove it from the building. I was using Pingtel phones that were supposed to "work great with Asterisk" but instead crashed all the time. Then I got the Sip bug that caused * to stop responding when doing a cvs update for work arounds for the phones. I ended up pulling the Pingtel phones and replacing with Smartalk analog phones. Everyone seems to be fairly happy right now, but it has been a huge nightmare and a huge learning experience. One thing I learned is that Asterisk is very capable even in production, but if your going to implement it you had better make sure that you can do full blown testing before implementing initially and again for any changes. I am very wary of CVS updates now... Quite a few people are making a concerted effort to make documentation better and I think this will help quite a bit. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan > Caruana > Sent: Thursday, October 16, 2003 8:21 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] I give up!! > > > i've just lost $2000 dollars or so on my first commercial asterisk > installation .. > i'm running a PIV class server, three Digium Wildcard FXO cards, and > 10 Grandstream Budgettone SIP phones. The system was to be a PBX > for a small company. After over 2 months of pissing about, the client has > had his fill of asterisk problems, and asked me to take my equipment > out of the building. Obviously, I haven't been paid for anything. > > The problems I faced were the following : > - initially a problem with asterisk crashing totally when there wasn't an > extension > to ring .. though this was fixed in a subsequent CVS, it was causing > downtime. > the client has no unix knowledge, and a script I put in to kick in the > asterisk > when it shut itself down didn't seem to always work. > > it also reduced the quality of my subsequent callout requests > to something > on > the lines of "the phone server is crashed again" regardless of what the > problem was > > - a dialplan problem, where one phone was ringing 10 seconds after the > others, >at the client's request and they were hearing other phones ring and > picking up >a non-ringing phone (ok, I can't really blame that on asterisk ..) > > - echo on the lines .. that after much fiddling around with configurations > went from >terrible to borderline acceptable. To people not used to digital > telephony and >computer stuff, the echo was VERY annoying. They used to avoid > the phones >because they said people would not understand them. > > - no consultative transfer. The closest I got was to park the > call, call the > other party, > tell him "a voce" which line the call is parked on and then get him to > pick up the call. > This is, in my opinion, a very basic feature that is missing on > asterisk. > The park/ > pick up sequence proved too difficult for the clients' secretaries to > grasp. > > - I could not get G729 working properly (license paid up, G729 up and > running). In > the absence of a manual, the fault solving process was > something like "ask > a question > on the mailing list, get a few answers, go to the client, try it out, > fail, go back home, > send another question on the mailinglist" with about 48 hours for each > iteration. I was > also appearing a real chimp "expermimenting" stuff at the > clients' office. > > At this point I decided to cut my losses, retreive the equipment > and call it > a day. > When asterisk is well documented and released in stable releases, I will > willingly > consider it again. I would be willing to pay for a stable, documented > version of > asterisk. It is a lovely software, and to begin with I was very > enthusiastic > about it. > I do understand that the support community is helpful, but the current > status of things > limits asterisk to a hobbyist scenario or at least somewhere > where there is > an engineer > with lots of linux experience and patience online 24 hours to > solve problems > as they > crop up. > > If anyone would like a couple of second hand FXO boards, contact > me. I have > already found a home for the grandstreams. > > cheers > Dave > > ___ > 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] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
> On the analog front, since we're talking about paging or intercom: it > has been mentioned that the Sayson ADSI phones (Aastra?) are > integrated with an Altigen PBX system, that the phones can support > paging. See the link below. > > http://www.sayson.com/product/Altigen.htm > > If this could be combined with a Cisco ATA-186 or similar product, > would it not be possible to have paging via SIP delivery? I have not > put anything other than the most basic thought into this, and I don't > know exactly how they support ADSI-based paging, but has anyone > worked on this? I mentioned it via IRC and someone said they took a > half-hearted stab at it, but it didn't sound like it had been > explored in depth. That is a major feature issue that most business > customers are looking for, and it would be great to have some > combination of devices that offered paging or intercom. > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users I have just put in some analog phones that offer intercom/paging and work rather well so far. They are Smartalk phones. (NOT ADSI) The guys there have been real good about helping me get buttons programmed exactly the way I want them etc. These phones also allow for staion monitoring and have an attendant console. These phones use 4-pair wiring so 3 lines + power pair. The power pair is also responsible for the intercom/paging/station monitoring. With the right wiring it might work with an ata device. Just a thought. http://smartalk.ca/nrgover.htm Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Andrew > Joakimsen > Sent: Friday, October 17, 2003 4:12 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer > for Cisco 7940/7960!!) > > > Which model? Are you using them directly with Asterisk? Analog phones > should only be 1 line, IMO all the call processing should be handled by > Asterisk. > I am using the ST310 and the SE310(console). The ST110,210,&310 are basically the same phone, its just a matter of how many lines the phone is programmed to take AFAIK. ie you could get an ST110 and later set up 2 more lines on it. As far as analog phones needing to be 1 line only, I have not seen a problem. It seems to complement * quite well. Because of the flexibility of * and these phones, the possibilies are interesting. Please correct me if I've missed something as to why analog phones should be single line. Andy > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Andy Hester > > Sent: Friday, October 17, 2003 10:11 AM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP > Auto-Answer > > for Cisco 7940/7960!!) > > > > > > > On the analog front, since we're talking about paging or intercom: > it > > > has been mentioned that the Sayson ADSI phones (Aastra?) are > > > integrated with an Altigen PBX system, that the phones can support > > > paging. See the link below. > > > > > > http://www.sayson.com/product/Altigen.htm > > > > > > If this could be combined with a Cisco ATA-186 or similar product, > > > would it not be possible to have paging via SIP delivery? I have > not > > > put anything other than the most basic thought into this, and I > don't > > > know exactly how they support ADSI-based paging, but has anyone > > > worked on this? I mentioned it via IRC and someone said they took a > > > half-hearted stab at it, but it didn't sound like it had been > > > explored in depth. That is a major feature issue that most business > > > customers are looking for, and it would be great to have some > > > combination of devices that offered paging or intercom. > > > > > > JT > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > I have just put in some analog phones that offer intercom/paging and > work > > rather well so far. > > > > They are Smartalk phones. (NOT ADSI) The guys there have been real > good > > about helping me get buttons programmed exactly the way I want them > etc. > > These phones also allow for staion monitoring and have an attendant > > console. > > These phones use 4-pair wiring so 3 lines + power pair. The power > pair is > > also responsible for the intercom/paging/station monitoring. With the > > right > > wiring it might work with an ata device. Just a thought. > > > > http://smartalk.ca/nrgover.htm > > > > Andy > > > > ___ > > 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] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Andrew > Joakimsen > Sent: Saturday, October 18, 2003 12:55 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer > for Cisco 7940/7960!!) > > > There is no rule, it is just my way of thinking. Everything related to > multiple lines should be handled by the Asterisk server / PBX. Is there > any specific advantage that you have seen to using multiline phones? > For the most part * does handle it all. However, having only 1 line makes it difficult to field calls (as in a receptionist). If you are on the phone & someone else calls, the Flash functionality no longer works. With 2 lines, a person could be on line 2, field a call on line 1 & do a consultative transfer without any problem. This is the main advantage. Also, you could make 3 3-way calls on this phone, assuming you had all 3 lines configured & live, and then bridge all 3 together to have a 6 person conference call from your phone. Most of my phones have only 1 line configured, but all of the secretaries have 2 lines. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of John Todd > Sent: Sunday, October 19, 2003 6:40 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer > for Cisco 7940/7960!!) > > > > > On the analog front, since we're talking about paging or intercom: it > >> has been mentioned that the Sayson ADSI phones (Aastra?) are > >> integrated with an Altigen PBX system, that the phones can support > >> paging. See the link below. > >> > >> http://www.sayson.com/product/Altigen.htm > >> > >> If this could be combined with a Cisco ATA-186 or similar product, > >> would it not be possible to have paging via SIP delivery? I have not > >> put anything other than the most basic thought into this, and I don't > >> know exactly how they support ADSI-based paging, but has anyone > >> worked on this? I mentioned it via IRC and someone said they took a > >> half-hearted stab at it, but it didn't sound like it had been > >> explored in depth. That is a major feature issue that most business > >> customers are looking for, and it would be great to have some > >> combination of devices that offered paging or intercom. > >> > > > JT > > > >I have just put in some analog phones that offer intercom/paging and work > >rather well so far. > > > >They are Smartalk phones. (NOT ADSI) The guys there have been real good > >about helping me get buttons programmed exactly the way I want them etc. > >These phones also allow for staion monitoring and have an > attendant console. > >These phones use 4-pair wiring so 3 lines + power pair. The > power pair is > >also responsible for the intercom/paging/station monitoring. > With the right > >wiring it might work with an ata device. Just a thought. > > > >http://smartalk.ca/nrgover.htm > > > >Andy > > Looks useful, but requires essentially a second line to work as a > pager or intercom. Not necessarily a bad thing, but as an an example > it would require a whole ATA-186 to just get one line and the > "paging" feature working, plus perhaps some additional wiring to work > that all into the power pair. Also, it is unclear if that will work > at all, since there is no documentation on their website about how, > exactly, the "pager" or "intercom" features work. It's completely > undiscussed at the nuts-and-bolts level (though they tell you what > buttons to push.) Snip > JT It runs on Cat5. The paging/intercom/station monitoring all occurs over the power pair. (pin1&8)The power pair wouldn't go to * at all. It works seemlessly so far on a channel bank. As for the ATA-186, I am not familiar with the wiring of the ATA, but it seems that you could use a surface mount jack with 1 or 2 lines as needed. You'd just have to get pair 1&8 hooked up to the central power supply. It may be more trouble than its worth to you, but I mentioned it in response to your earlier post just because people always seem to be asking for these features regarless of the type of phone. Andy > ___ > 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] zhone z-plex 10
Is there a trick to getting into a zhone z-plex 10 through the serial interface? I tried using a couple of terminal programs the other day and didn't get a login. I have a port that quit working and it doesn't appear to be wiring related. I had another case like this the other day, and after fiddling with the wiring for a long time, I was ready to give up for the night. I restarted everything one last time for grins and the port came back up. Now I have a different port down and I am wondering if the z-plex is downing the port for some reason or other. Thanks, Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Panny > Malialis > Sent: Wednesday, October 22, 2003 11:43 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Meetme > > > Is app_meetme broken? > > I seem to get invalid conference number all the time :( > > Panny I was having this problem and found out that you can't change the context for meetme HTH Andy > ___ > 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] Call transfering, conferencing
> Peter Hudec wrote: > > > hello, > > > > my questns are about few * functionality. > > > > 1) how can I make call tranfer. Not call parking. > > If I'm talking with some one a I want to tramnfer call to the another > > extension, to the other person. > > > > 2) how can I make call confernece. Not Meetme > > If I'm talking with some one and I want to join another person to our > > talk . > > > > I haven't found this in any manual :( > > > > hudecof > > > You won't find in in any Asterisk manual becasue these are not features > of Asterisk, they are features on the phone.. The phone needs to support > transfer and if you want conferencing without using "meetme" then you > need a phone that supports conferencing.. > > Later.. These features are supported on analog lines. Transfer can be done by initiating a 3way call and then hanging up. This works both blind and consultative for me. 3 way calls are initiated with a Flash. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie hardware question
-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Just MESent: Thursday, October 30, 2003 11:00 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Newbie hardware question Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got "drafted" to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a couple of spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using because I will run out of PCI slots if I use 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks Jon Hoffman Jon, Steven just answered this question quite well, so I'll just refer to him: Andy snip You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. The Adit cards are 8 lines at a time. The Adtran unit would let you get 18 extensions and 6 incoming lines on a single T1 interface. Both of these units can be bought on Ebay for relatively inexpensive compared to new prices. Then you will either have to scour the net for the FXO card, or go pay full price for it. Either way, this gets you down to 1 PCI card. If you go the route of a T400P card, adding more service later will be less of a hassle. You could also use it to do your network routing if you decide to go frac T1 for data and some phone service tacked onto the same T1 interface. This could potentially even be a better route as you wouldn't need to find FXO interfaces anymore. You would also get the benefit of using the new software fax setup to get yourself on the way to unified messaging. -- Steven Critchfield <[EMAIL PROTECTED]>
[Asterisk-Users] Strange Problem with Asterisk....
Wondered if anybody might have some ideas about what could be causing this I have a T100p hooked up to an Adit 600 with 12 channels of voice off of a T-1 coming in. I have a t100p connected to a zhone z-plex with 24 fxs going to my stations. Some of the station are 2 line phones. These have 2 zap channels that are dialed when the extension is matched. Also, some extension require ringing 2 phones at the same time. For example the boss wants his assistants phone to ring also when someone calls him. Then the calls roll to a receptionist if not answered. Finally the call goes to voicemail This has been working fine for weeks now, but today the customer told me that one of the phones configured this way was hearing 3 to 4 other conversations on their phone. The phone did not give dialtone, but in picking up the handset you could hear someone shuffling paper working at their desk etc. When certain other extensions were on calls, you could hear clearly the person they were connected to, but not the person in the office. I checked all the wiring and couldn't find any problem. I rebooted the channelbank thinking maybe somehow it had gotten screwy. No change. Then, just for grins, I stopped and restarted *. Voila, the problem was gone. Can anyone think of a scenario that could cause * to do this? Thanks, Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange Problem with Asterisk....
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer > Sent: Thursday, November 06, 2003 10:46 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Strange Problem with Asterisk > > > I think there are issues with combining flash-hook supervised transfers > with meetme conference bridges. Can you find out if that took place, i.e. > someone tries to transfer into a meetme conference? > > Mark No, I have Meetme on an extension, but the users don't even know about it and the ext# for it is in a completely different range. I don't know if I explained it adequately looking back on my post. The situation persisted over 8 to 10 hours and through numerous calls. what seems weird to me is that the other extensions functioned correctly, but the one didn't. I could pick up the handset and be listening to the sounds of someone working quietly at their desk, and then if they made a call, I could hear the ringing, the person answer, but not the caller. this happened over and over. But it didn't pick up the audio of all the other zap channels, just 4 or 5. Thanks, Andy > > On Thu, 6 Nov 2003, Andy Hester wrote: > > > Wondered if anybody might have some ideas about what could be causing > > this > > > > I have a T100p hooked up to an Adit 600 with 12 channels of > voice off of a > > T-1 coming in. > > I have a t100p connected to a zhone z-plex with 24 fxs going to > my stations. > > > > Some of the station are 2 line phones. These have 2 zap > channels that are > > dialed when the extension is matched. > > Also, some extension require ringing 2 phones at the same time. > For example > > the boss wants his assistants phone to ring also when someone calls him. > > > > Then the calls roll to a receptionist if not answered. > > Finally the call goes to voicemail > > > > This has been working fine for weeks now, but today the customer told me > > that one of the phones configured this way was hearing 3 to 4 other > > conversations on their phone. The phone did not give dialtone, but in > > picking up the handset you could hear someone shuffling paper working at > > their desk etc. When certain other extensions were on calls, > you could hear > > clearly the person they were connected to, but not the person > in the office. > > I checked all the wiring and couldn't find any problem. I rebooted the > > channelbank thinking maybe somehow it had gotten screwy. No > change. Then, > > just for grins, I stopped and restarted *. Voila, the problem was gone. > > > > Can anyone think of a scenario that could cause * to do this? > > > > Thanks, > > Andy > > > > ___ > > 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] Help with Warnings
I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be appreciated as I'm not sure of the cause
/solution.
Here are the errors:
Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
(zt_call): cidspill already exists??
+
/* Don't send audio while on hook, until the call is answered */
p->dialing = 1;
if (p->use_callerid) {
/* Generate the Caller-ID spill if desired
*/
if (p->cidspill) {
ast_log(LOG_WARNING, "cidspill
already exists??\n");
free(p->cidspill);
}
++
Nov 17 17:13:51 NOTICE[1242768320]: File app_dial.c, Line 502
(dial_exec): Unable to create channel of type 'Zap'
++
/* Request the peer */
tmp->chan = ast_request(tech, chan->nativeformats,
numsubst);
if (!tmp->chan) {
/* If we can't, just go on to the next call */
ast_log(LOG_NOTICE, "Unable to create channel of
type '%s'\n", tech);
if (chan->cdr)
ast_cdr_busy(chan->cdr);
free(tmp);
cur = rest;
continue;
}
++
Nov 17 17:20:57 NOTICE[1209214400]: File chan_zap.c, Line 3462
(zt_read): Fax detected, but no fax extension
++
/* Fax tone -- Handle and return NULL */
if (!p->faxhandled) {
p->faxhandled++;
if (strcmp(ast->exten, "fax")) {
if (ast_exists_extension(ast,
ast->context, "fax", 1,
ast->callerid)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_
PREFIX_3 "Redirecting %s to fax extension\n",
ast->name);
/* Save the DID/DNIS when we
transfer the fax call to a "fax"
extension */
pbx_builtin_setvar_helper(as
t,"FAXEXTEN",ast->exten);
if (ast_async_goto(ast,
ast->context, "fax", 1, 0))
ast_log(LOG_WARNING,
"Failed to async goto '%s' into fax of
'%s'\n", ast->name, ast->context);
} else
ast_log(LOG_NOTICE, "Fax
detected, but no fax extension\n");
} else
++
Nov 17 17:26:04 WARNING[1234379584]: File chan_zap.c, Line 3331
(zt_read): zt_rec: Unknown error 500
+++
/* Check for hangup */
if (res < 0) {
if (res == -1) {
if (errno == EAGAIN) {
/* Return "NULL" frame if there is nobody
there */
ast_mutex_unlock(&p->lock);
return &p->subs[index].f;
} else
ast_log(LOG_WARNING, "zt_rec: %s\n",
strerror(errno));
}
ast_mutex_unlock(&p->lock);
return NULL;
+
thanks,
Andy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
Just a had to put in a few points on this... First, it is correct that there is no cause to be rude, either by repling rudely or posting without doing any research. I think that a response directing them to the proper resources is better than not responding at all. Second, one of the main problems has been documenatation as everyone knows. As one of the people suggesting a wiki several months back, I am thankful to those who have hosted/maintained/posted. Searching the mailing list archives can be futile in a lot of cases because it can be to tedious/laborious to find an answer in a timeframe that is practical. This is why we need the wiki. I would suggest that we start refering them to the wiki as well as the mailing list. Props to Olle & BKW for responding with their docs. Lastly, I'm not sure that the footer idea will work at all. It is doubtful that the people asking the questions in question will read the footer. The idea is to put links to the documentation, wiki, unofficial * pages and instructions BEFORE the mailing list stuff on the Asterisk support page. Other wise many will not even see it much less take the time to read it. I believe Critch suggested something like this in a thread a few days ago. ie you can only post after you've read the instructions or something. Snip With the exception of I don't know how hard it is to setup, I wouldn't mind this going to a semi moderated group. RO access requires little intervention. Basically it is the default. Posting requires a quick read of the FAQ with a quick push through a small and to the point netiquete page, and then maybe a 2 or 3 question pop quiz afterwords. After that, release the posting restriction. It is fairly minimalistic, and shouldn't get too in the way of users who want to lurk and read first. Snip We as a community have made great strides from even a few months backas far as docs goes, I think we just need to make sure it gets out there and then if people still ask questions without research, we can turn Critch loose on 'em. ;) Sincerely, Andy Hester ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
I agree, seperate digium list is best. Mainly because it will help build business for consultants who could produce growth for Digium. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of John Todd > Sent: Thursday, November 20, 2003 9:59 AM > To: [EMAIL PROTECTED] > Subject: Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business > discussion again) > > > > > So far it seems like the proposed candidates for new lists are: > > > >> asterisk-newbies (perhaps a better word?) > >> asterisk-nontech > >> asterisk-biz > > > >> The amount of mail on asterisk-users is more than even *I* > can read in a > >> day, and my job is 100% asterisk. There probably is a > justification for > >> a new list, but I think it is less the -biz list as much as > much as the > >> -newbies. Keeping a business discussion on -users is probably quite > >> useful since often times a business discussion can involve technical > >> details of what Asterisk is capable of doing. > > > >Agreed... biz is just a special class of users, but what would go in > >nontech... newbies wouldn't get much traffic since nobody wants > to really > >admit they're a newb and moreso they'd get frustrated that the people who > >really do know wouldn't hang out there. > > > >Although I do like -biz on a separate list because you can also see who's > >offering what, and get help on how to set it up and interop -- > think of all > >the vonage, nuphone, p8, ich and other "how do I do this" traffic we've > >seen on -users lately... > > > >Ugh. I hate trying to figure things like this out. :-) > > > >Regards, > >Andrew > > I have no opinion on the "newbies" and "nontech" lists, but I > strongly favor a "biz" list, since I have held off on many occasions > from posting "I need a provider in X city who can terminate via IAX" > or "I need a set of asterisk-clued hands in X city" because I knew > that quite a few people (mostly "businesspeople") would use the > "reply-all" feature to spam the list with their replies which should > be to me personally. > > I believe these should be digium-sponsored lists, due to the fact > that I'd like to keep the focus of the project on Digium's resources, > to help drive business into their card and device sales projects. > > JT > ___ > 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 Lists (was Re: [Asterisk-Users] Asterisk Businessdiscussion again)
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Thursday, November 20, 2003 2:19 PM > To: [EMAIL PROTECTED] > Subject: RE: Asterisk Lists (was Re: [Asterisk-Users] Asterisk > Businessdiscussion again) > > > On Thu, 2003-11-20 at 12:06, Andy Hester wrote: > > I agree, seperate digium list is best. Mainly because it will > help build > > business for consultants who could produce growth for Digium. > > The problem with a list for this type of information is that a new > subscriber will not have information in front of them quickly unless > people are answering the questions. The information about businesses > that support asterisk either through installs or service is for the most > part static and better suited to maybe a few pages on the Wiki. > > After that, most other business related questions could easily be > handled here. > > I admit I may be blind concerning some aspect of discussion you have > felt shouldn't take place on this list, or couldn't be addressed by a > web page tightly linked with the documentation. If so please enlighten > me so we are on the same page. > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users Maybe I'm going about this all wrong. I was really thinking more about development of the Digium channel for those who are trying to sell the product/services. It seems to me that most of the subsribers by far are the do it yourself types anyway and would only pay as a last resort. This is fine but not a huge market for consultants. I was thinking more about discussions of how to break into certain market segments, where is asterisk a good fit, how have I done this that or the other. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
> > We as a community have made great strides from even a few > months backas far > > as docs goes, I think we just need to make sure it gets out > there and then > > if people still ask questions without research, we can turn > Critch loose on > > 'em. ;) > > I really don't want to be the attack dog. I've pointed out that I am > doing my best to relieve my aggression before coming online to answer > messages. My metalworking is coming along very nicely. Just beware when > my metalworking starts turning to weapons instead of artwork and > furniture. I should be starting my car shortly enough. Then beware of > the 500cid sportscar with the cow catcher on the front. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users Please send pictures! Can I get One? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAX, IAX2 and latency
Peer, Would you be interested in helping me test diax from Germany to my * box here in Texas? I just want to see about latencty etc.. If so email me offlist so I can set up an extension/registration for you. I am working up a proposal for a conferencing server for a customer whose main office is in Hamburg. I hope to be able to use his existing ADSL line and use DIAX or Hard phones to connect to the * server here. I will also get voip 800 service for others who might need to call in to the conference. The conferences right now are only 4 or 5 people. Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Peer Oliver > schmidt > Sent: Friday, November 21, 2003 11:50 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] DIAX, IAX2 and latency > > > Hello, > > today I tried a DIAX -> * -> DIAX connection over the internet (768/128 > ADSL connection on both sides). > > The sound quality was great. However, we had some latency problems, and > also, if both sides where not talking the first words had some problems > getting thru. > > Is this expected, is there anything that can be done on our setup, any > magical iax.conf entry? > > Thanks and best regards from Germany > > pos > > > > ___ > 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] MOH - Hold Button - I think I'm going crazy
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of PBX > Sent: Friday, November 21, 2003 6:33 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy > > > Ok... I know I have asked this question before, but have never gotten an > answer... When I press the hold button on my phone, should the caller > hear music just like when I park the caller or transfer them to another > extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker & mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy > > Please assist... > > -gcc > ___ > 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] MOH - Hold Button - I think I'm going crazy
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of PBX > Sent: Saturday, November 22, 2003 1:00 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going > crazy > > > Is there a solution to have the hold button to play MOH. Or even some > type of ADSI function that allows for this? > > -gcc There is a seperate module for the phones I use that will do Music on hold for all of the phones in the group. These are not ADSI phones, I guess you might call them analog feature phones. I don't think that Flash is a very good way to effect hold, since it is also used for other functions, so I guess the answer is that it probably won't work like you want it to without some changes. HTH Andy > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester > Posted At: Friday, November 21, 2003 10:17 PM > Posted To: Asterisk User Group > Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going > crazy > Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going > crazy > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of PBX > > Sent: Friday, November 21, 2003 6:33 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy > > > > > > Ok... I know I have asked this question before, but have never gotten > > an answer... When I press the hold button on my phone, should the > > caller hear music just like when I park the caller or transfer them to > > > another extension? > > It depends... If the button uses a function of the phone to > hold the call(ie keeps the call active but mutes the speaker & mic),then > you will not hear music on hold from *. I have some phones that can > operate this way. > > Andy > > > > > > Please assist... > > > > -gcc > > ___ > > 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol > Sent: Monday, November 24, 2003 5:46 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] New DIAX - version 0.9.4 - a big step > forward - available for download > > > Dan, > > I have been working with the new version and have discovered a strange > issue. When I place a call from a DIAX phone, all seems to generally > work properly. However, calls placed from either of my SIP agents > (Grandstream 101 hardphone and X-PRO softphone) or from the PSTN via > X100P often fail to connect. I had a similar experience when I upgraded to 0.9.4. i was running a cvs version from late october. blew away my src tree and cvs checkout resolved it. Maybe something related to IAX2. HTH, Andy > > This is especially true if DIAX has previously received a call, and that > call has been disconnected from the remote end. > > I have watched the Gastman (graphical asterisk manager) application, and > my own Call Manager application and both indicate that the system _is_ > ringing the DIAX channel. The DIAX just never rings, or changes to a > state where the user can answer the incoming call. > > A few experiments: > > I. > When the DIAX hangs up the call (is the _first_ party to hang up) the > call is terminated cleanly in all tests. When the remote party (the GS > or XTEN) breaks the connection, the GS does not always drop the > connection immediately (though the Manager shows the call being torn > down and both parties dropping out). When this occurs the DIAX will not > ring another incoming call -- it requires a restart first. > > II. > This does not always happen. Sometimes the DIAX will stop responding > even if it properly hangs up. Also, sometimes it takes several calls > before DIAX stops responding. > > III. > Even if DIAX has stopped responding to incoming calls, it is still > capable of making outbound calls, apparently without any problem. > > IV. > Even when DIAX doesn't immediately detect the handup (or respond to the > hangup event) and it is manually hung up after the call is torn down, it > is _sometimes_ still capable of receiving incoming calls. > > Could this have something to do with my configuration? For what it's > worth, I don't see any errors in the Asterisk log. > > 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] Asterisk and Avaya IP phones
Title: Asterisk and Avaya IP phones -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Ed RubrightSent: Thursday, December 04, 2003 5:03 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk and Avaya IP phones The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it’s a SIP phone or not. Thought maybe someone on this list would know. Question: Would I be able to register my Asterisk server or an individual SIP phone (Cisco 7960 or Polycom IP600) with the Avaya server these 46xx IP telephones use? I don't know what model of the Avaya server the company has purchased, so I have limited info here. Thanks in advance, = Ed Rubright These phones are H.323 phones from what I remember of the documentation, so they are compatible in a VERY general sense. However, I'm sure that there are alot of proprietary things that the phone does that would have to be sniffed out in order to make it fully compatible if that is even possible. I wouldn't expect any help from Avaya either. :) HTH Andy
RE: [Asterisk-Users] Port density: DS3 cards?
> > The data-only cards for DS3 seem to be in the "reasonable" price > range, though I have _no_ idea if they could be turned into > TDM-capable cards. Examples that were shown to me: > > http://oem.imagestream.com/PCI_720.html > http://www.ace-electronics.com/Hardware/T1E1J1/wanPCI-1T3.html > > A little more time with Google perhaps would discover other > solutions. These are, from what I gather, very inexpensive devices > in the grand scheme of things, and I believe some already offer Linux > drivers (though no mention of open source that I could find, I > imagine that these companies will be all over opening up more markets > for their cards.) > > Of course, Digium could keep it's leadership and our (collective) > money by starting to poke around at such a driver or card. It's > really a chicken-egg situation: nobody will want to muck with driver > authorship or card production until there are buyers, and there won't > be any buyers of such "experimental" technology unless it's cheap to > experiment with, just like the T100P cards are. Open source is still > scary to bell-heads, and they will resist until they actually see > (with their own eyes) a working system that replaces their $100k > CisNorSiemAvaytelensaco boxes with a $7k PC/card combination. Even > then, it's still an uphill battle, but at least it's a battle, > whereas right now it's a complete non-starter to open one's mouth > about open source telephony gatewaying at truly large scale > installations. And, to be honest, the telco guys are correct at this > moment. > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users I have been mulling over what it would take to get drivers done for ImageStream's products. They have a component architecture that is supposed to reduce development time/cost. The component stuff is open source. The part of the driver that you have to write can be open source or proprietary. I am not much of a coder, but someone more knowledgeable may be able to do it without too much trouble. I am an ImageStream reseller - if you need hardware I'll give you good pricing. ;) Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Port density: DS3 cards?
> >I have been mulling over what it would take to get drivers done for > >ImageStream's products. They have a component architecture that > is supposed > >to reduce development time/cost. The component stuff is open > source. The > >part of the driver that you have to write can be open source or > proprietary. > >I am not much of a coder, but someone more knowledgeable may be > able to do > >it without too much trouble. I am an ImageStream reseller - if you need > >hardware I'll give you good pricing. ;) > > > >Andy > > > Shoot, set me up with 42 2u servers > with dual TE410P boards, and then 12 M13 muxes, and then 1 12-port > DS3-to-OC12 mux (or 3 DS3-to-OC3 muxes, and one 3 port OC3-to-OC12 > mux) and we can even test one of those OC12 boards that ImageStream > sells! > > Why don't you ping someone at ImageStream and see if they're willing > to offer a DS3 developer kit for some interval (6 months? 8 months?) > to a developer if they show appropriate interest and expertise. > Anyone want to volunteer? > > Actually, I'd ask a senior developer at ImageStream to see if they > think it's even possible first; they'll at least be able to say if > it's in the realm of sanity. You have the inside track; let us know > what you hear. > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users I'll follow up on this tommorrow and let you know what I hear Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Port density: DS3 cards?
> > > >I have been mulling over what it would take to get drivers done for > >ImageStream's products. They have a component architecture that > is supposed > >to reduce development time/cost. The component stuff is open > source. The > >part of the driver that you have to write can be open source or > proprietary. > >I am not much of a coder, but someone more knowledgeable may be > able to do > >it without too much trouble. I am an ImageStream reseller - if you need > >hardware I'll give you good pricing. ;) > > > >Andy > > > Shoot, set me up with 42 2u servers > with dual TE410P boards, and then 12 M13 muxes, and then 1 12-port > DS3-to-OC12 mux (or 3 DS3-to-OC3 muxes, and one 3 port OC3-to-OC12 > mux) and we can even test one of those OC12 boards that ImageStream > sells! > > Why don't you ping someone at ImageStream and see if they're willing > to offer a DS3 developer kit for some interval (6 months? 8 months?) > to a developer if they show appropriate interest and expertise. > Anyone want to volunteer? > > Actually, I'd ask a senior developer at ImageStream to see if they > think it's even possible first; they'll at least be able to say if > it's in the realm of sanity. You have the inside track; let us know > what you hear. > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users I talked to Imagestream this morning about the possibilites. Their lead engineer said that there would be no way to do voice over their DS-3 cards using software processing because it would take too much processing power. It would be possible to do some custom design for their boards that incorpotates hardware processing, but he doesn't know of anything currently available. So unless there's something I/he missed, I guess the answer is no on the DS-3. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Port density: DS3 cards?
> > > >I talked to Imagestream this morning about the possibilites. Their lead > >engineer said that there would be no way to do voice over their > DS-3 cards > >using software processing because it would take too much > processing power. > >It would be possible to do some custom design for their boards that > >incorpotates hardware processing, but he doesn't know of > anything currently > >available. So unless there's something I/he missed, I guess the > answer is > >no on the DS-3. > > > >Andy > > I have no reason to disbelieve this report, but I will offer some > minor scepticism at this reply. A well-equipped PC can currently > handle 8 T1 channels, and it seems that only the IRQ issue is causing > more channels to not be viable in the current TE410P environment. It > would seem reasonable to think that a very well equipped PC (4-way, > 8-way?) would be able to handle the "processing power" requirements > of a DS3, whatever was meant by that statement. Of course, there may > be other underlying issues specific to ImageStream that make this > impossible; I don't know. > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users The guy did leave open the possibility that he could be wrong, and said that he'd be glad to answer any further questions or if we had some other way of doing it. If you or some of the others think that this should be possible then perhaps we could get together a list of more specific questions to ask. Thoughts? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO cards
Comments Inline > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sri > Sent: Tuesday, December 09, 2003 4:41 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] FXO cards > This maybe a stupid question. Pardon me. > I see everyone talking about purchasing the channel bank from ebay. > 1. As a user who has never used ebay, are these used equipments ? Some are, Others are not. > 2. Are these reliable in terms of all ports working and all hardware intact? It depends on the auction. There are no gaurantees, but most people will either list any known problems or specifiy that the unit is sold as is. > 3. Is there a huge price difference between purchasing it from a authorized dealer and ebay ? Most of the time > 4. Or was this suggestion just because the system being setup for is a NPO who like to save, even if it is a couple of dollors? Not sure. > 5. If we find some issues with it, can we return it for another ? Could be, but there's always a chance of getting stuck with it > 6. How far reliability becomes an issue in purchasing it from ebay or an authorized reseller. Once again, most people have good experiences or ebay wouldn't do so well, but ther are no gaurantees. :) > Cheers > Sri I think ebay is fine. I also think there are times to buy new equipment. It really depends on the circumstances and the needs of the person. I guess it comes down to what you feel comfortable with. Sincerely, Andy Hester ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Transfer/Voicemail Bug
Hi All, Is anyone working on bug id 617? http://bugs.digium.com/bug_view_page.php?bug_id=617 I am not sure where to start and I really need to get this fixed. Any ideas? Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zhone Zplex Hack & Diddling the Battery
I came across this hack while looking through chan_zap.c. Is this still applicable for those w/ zplex cb's? Is it a maybe need or an absolutely need? Any background info would be appreciated. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax detection: false positive
john lawler wrote: Hi guys, I just moved from Asterisk release 0.5.0 to CVS 2003-12-22, and after overcoming a few changes in my configuration, I encountered one problem that I couldn't shake that was working fine in 0.5.0. It's the fax detection. I just have a simple extension setup like this: exten => fax,1,Dial(Zap/4,30,tr) exten => fax,2,Hangup in my main incoming context. This used to work fine, I don't think I ever had a false positive or negative, but now just about every call (possibly every call) that comes in when I've got that extension defined rolls to my fax machine on Zap/4 immediately. I'm sure others have encountered this and I missed a post on some change to the configuration that might be causing this. Thanks for your help. jl The issue is in dsp.c. Check out an old copy of dsp.c and replace the one in your source tree. Then recompile. I am using the file from 12/08/03, as someone noted that this one works. ajh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Single/Dual DS3 - anyone seen this?
Andrew Kohlsmith wrote: http://www.imagestream.com/PCI_720.html Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andrew, I talked to the tech at Imagestream about their products a month or two ago when there were discussions on the list. The tech was fairly insistant that the zap divers must be making use of voice processing on the zap boards and that it would be impossible to do ds3 with all processing on the main cpu. I thought that all of the processing was done on the cpu for the zap cards but I don't know this absolutely. His suggestion was to use digium cards in Imagestream chassis and intergrate asterisk/zaptel into the linux distro on the unit. But of course this wouldn't get you DS3 :) Andy ___ 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] Pingtel Phones?
[EMAIL PROTECTED] wrote: Hello All, Does anyone here have any experience with pingtel Xpressa hard phones? I am considering buying a couple. Already have Snom200s, but want something with better CTI and full duplex speakerphone. Michael Michael, I used some Pingtel phones with * and ended up having to scrap them. Pingtel assured that they would work with * but when they didn't, they put it all off on Asterisk. They also didn't seem to be interested in getting them to work. In my opinion they were dishonest with through the whole deal and I wouldn't recommend doing business with them or using that phone. One persistent issue is that the phone would lock up if you had more that 2 calls even though they said it could handle 11 simultaneous calls. I talked to at least one other user who had the same issues. Everything was fine in the lab, but upon roll out the phones didn't work right. Andy ___ 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] Configuring Pingtel Xpressa
Michael Graves wrote: Well my Pingtel Xpressa arrived today. It's configuration is nowhere near as clear as the SNOM 200. Can someone here provide some guidance on getting the Xpressa talking to *? Michael Michael, In my experience, the Xpressa most times registers with * but never realizes that the registration is complete. This is eveidenced by the browser hanging forever in the Xpressa config when you try to set it to register.(can't remember their exact terminology for this setting but its kind of odd) Anyway, the result is that it never tries to reregister and therefore the phone comes unregistered at the end of whatever the registration period is set for and ceases to work. You can set Asterisk to use the phones unregistered and keep the phones set to not register and they will make and recieve calls. I will try to dig up my old notes and configs for you if you like. Andy ___ 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] Suggestion for a new asterisk setup.
Hello all, I need to setup a new asterisk system with the following requirements: 1. Will be moving from chan_sccp to sip (7960's), but I want to support the sccp phones until everyone has been migrated. 2. Need to maintain current portability of the 7960's. (ie a user can unplug his phone from the internal LAN, take it home or wherever, and plugin and have the phone register and work just like in the office. I have tried the following: 1. Asterisk server on LAN behind NAT, LAN phones on the same net, tested a phone with a public IP. 2. Asterisk on Public IP, LAN phones on LAN behind NAT, didn't even get to testing remote phones. I am having trouble with calls completing but not passing the audio stream. I have done fixup SIP/opened 5060/tried many settings in SIP.conf/set 7960's to NAT=YES etc. I can not NAT each phone individually and allow RTP to it, as I saw one person did. I really don't want to run asterisk on a public IP and a LAN IP going around my firewall. Do I need to put the phones on a separate LAN network and run asterisk on a public ip and private? Do I need to run a SIP proxy. I looked at SER/OPENSER, but it seems to break some things. (Need to be able to record all calls need MWI) Should I run 2 asterisk boxes connected with maybe TDMoE? Would that work? Any suggestions would be greatly appreciated. Thanks, Andy Hester ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Suggestion for a new asterisk setup.
Andrew, Thanks, for the response. That is a very clean solution and much less work/complication, however, I am not sure that the security guy for this network will allow me to put up the asterisk box dual homed to the public IP and the LAN. If there is not another feasible way then I may end up going with this anyway. Any other feasible ways to accomplish this? Sorry for the top post... Having to use Outlook for the moment. Thanks, Andy Hester From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Friday, January 12, 2007 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Suggestion for a new asterisk setup. I assume there is one NAT router for the LAN and nothing fancy, so setup the Asterisk machine on the router/firewall (or make it such) and have it listen on both LAN and WAN interface. Now use a hostname for the SIP server, and run a DHCP/DNS server that will resolve that hostname to the LAN IP address of your router, when it is queried from the LAN side, when from the WAN side it would just be the regular lookup (use FQDN). Now phones will work from anywhere, no NAT issues to deal with at all. Each interface that asterisk runs on is isolated from the other. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Suggestion for a new asterisk setup.
In the current setup, asterisk is behind a different nat/firewall than the LAN phones. The phones are using sccp. If the asterisk box is compromised, it is not on the local LAN. This is what I think he doesn't want to give up. Andy > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Colin Anderson > Sent: Friday, January 12, 2007 12:20 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users] Suggestion for a new asterisk setup. > > >I am not sure that the security guy for this network will allow me to put > up the asterisk box dual homed to the public IP and the LAN. > > Your security guy needs to go back to school. If eth0 is on the LAN and > eth1 > is on the WAN, and the WAN connection is properly secured with only the > ports you need, and your SIP passwords arent 1234 or something that can be > guessed, what difference is there between this configuration and port > forwarding? The footprint you are exposing to the public internet is > exactly > the same. The only thing that I can think of is for IDS, you may have a > firewall that does this. Optionally, one could run a "soft" firewall on > the > WAN side that supports IDS if that is the issue. Otherwise, why not? > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Trunk timing
I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not configured correctly, but I just wanted to make sure as I can't seem to find any reason at the moment. Thanks, Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX Trunk timing
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Zoa > Sent: Tuesday, January 16, 2007 11:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] IAX Trunk timing > > > You need a timing device on both ends. > > Zoa > But ztdummy should suffice yes? Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
I am following the example at
http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no
matter what, the call is connected. Can anyone confirm that config is working
for them? Any suggestions appreciated.
I need to transfer calls to a list of cell phones, ring all of them, allow them
to screen the call, connect the call to the first number that accepts the call,
and allow others to reject the call.
Thanks,
Andy
[macro-screen]
exten => s,1,Wait(1)
exten => s,n,Background(csp_ackshort-male)
exten => s,n,Set(TIMEOUT(response=10))
exten => 1,1,NoOp(Call Accepted)
exten => 2,1,Set(MACRO_RESULT=CONTINUE)
exten => t,1,Set(MACRO_RESULT=CONTINUE)
;original macro
;exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
connect the caller
;exten => i,1,Set(MACRO_RESULT=CONTINUE)
;exten => t,1,Set(MACRO_RESULT=CONTINUE)
[default]
include => architel
include => local
include => trunkintl
exten => _6XX,1,Dial(ZAP/G1/${EXTEN:1},40,M(screen))
exten => _6XX,2,Hangup
[findme]
exten => s,1,Playback(transfer_csp-male)
exten => s,n,Dial(LOCAL/6${findme1}&LOCAL/6${findme2},40,m)
exten => s,n,Hangup()
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RE: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
-Original Message- From: [EMAIL PROTECTED] on behalf of Philipp Kempgen Sent: Sun 4/1/2007 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT Andy Hester wrote: > exten => s,n,Set(TIMEOUT(response=10)) Should be exten => s,n,Set(TIMEOUT(response)=10) Regards, Philipp -- Thanks Philipp, I fixed this, but I still have the problem that the call is connected regardless... -Andy <>___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
Andrew Joakimsen wrote: > The logic of the macro is totally opposite of what it should be. I do > recall sending a corrected version of the script to someone a while > back, it might be on the mailing lists archive. > > However, there is an option for the Dial() command to do exactly what you wish > > p: This option enables screening mode. This is basically Privacy mode Thanks for the response - I missed that Dial option... Couple of questions on this: 1. I do not want to screen based on caller, instead I need to play the same message to a list of potential call recipients and allow each recipient to decide whether or not to accept the call based on whether or not they are available (for work for example). I understand that this option checks for a file. I will be transferring a call to this call coverage. How do I make sure that all the calls look for the same recording to play to the call screeners? 2. Does anyone have any dial plan examples of this type of set up? Thanks, -- Andy Hester Network Engineer Architel <>___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
