RE: [Asterisk-Users] Polycom 600 as a Receptionist Phone
Well, the phone automatically does call waiting on each line you register so you will be able to get a call on each line. You could always do this In Asterisk Setup extensions 100, 101, 102, 103, 104, 105, 106 Set your dial plan to ring 100 on all incoming calls. Set 100 to roll through 101 through 106 until it hits an open line. In phone... The label you use on the display is selectable so you can say it is one number but register it as another. Register 101 - 106 on the phone but name them whatever number you want. All lines appar the same at both ends now. How about that? W -Original Message- From: Brian Pavane [mailto:[EMAIL PROTECTED] Sent: Monday, November 08, 2004 3:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom 600 as a Receptionist Phone I am attempting to setup a Polycom SoundPoint 600 in the same manner that I have a Cisco 7960 (SIP) operate as a receptionist phone. With the Cisco 7960, I am able to have 6 line appearances all display the same phone number, and thus give the receptionist the ability to handle 6 simultaneous calls. I would like to do this same setup with the 6 line appearances on the SoundPoint 600, and thus give the receptionist the ability to select lines via the line apperance buttons, and handle her active calls via the line appearnce buttons. With the Cisco, I can simply take the same lineX settings, and reproduce them up to 6 times, and thus have the phone handle up to 6 appearnces of the same line. When I have attempted to do the same thing with the Polycom configuration, I was not able to get this to work. Can anyone provide a Polycom configuration file for a receptionist phone? Thank you. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI Doesn't Turn Off
From my voicemail.conf, my context where I define my mailboxes in this file is [sip] From: Paul Rodan [mailto:[EMAIL PROTECTED] Sent: Monday, November 08, 2004 2:49 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] MWI Doesn't Turn Off What is the [context] you are using in voicemail.conf ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. SilerSent: Monday, November 08, 2004 3:48 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] MWI Doesn't Turn Off Interestinging From my voicemail.conf, my context where I define my mailboxes in this file is [sip] In the sip.conf I have [EMAIL PROTECTED] Changed that to [EMAIL PROTECTED] and it seems to work better now. Thanks! Wiley From: Paul Rodan [mailto:[EMAIL PROTECTED] Sent: Monday, November 08, 2004 12:58 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] MWI Doesn't Turn Off What does it show in /var/spool/asterisk/voicemail/default/extension/INBOX/ ? Sometimes when my users delete a message or move them around, the sequential order in the INBOX will get thrown off. So the phone’s light will stay on, because Asterisk can see a file(s) in there, but when they go to access their voicemail, it’ll say they have no messages, because the voicemail system doesn’t see a msg0.wav file, instead there would be a msg6.wav file or something like that in there. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. SilerSent: Monday, November 08, 2004 2:20 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] MWI Doesn't Turn Off Anyone having issues with the message indicator lights after CVS-HEAD-07/23/04-13:55:59 ?? For several of my users, our MWI lights do not turn off. Phones are Polycom IP500 and this just started prior to my last update. Should I update to a newer version? I pulled this from the CVS last week so I thought it was newest. Thanks, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI Doesn't Turn Off
Interestinging From my voicemail.conf, my context where I define my mailboxes in this file is [sip] In the sip.conf I have [EMAIL PROTECTED] Changed that to [EMAIL PROTECTED] and it seems to work better now. Thanks! Wiley From: Paul Rodan [mailto:[EMAIL PROTECTED] Sent: Monday, November 08, 2004 12:58 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] MWI Doesn't Turn Off What does it show in /var/spool/asterisk/voicemail/default/extension/INBOX/ ? Sometimes when my users delete a message or move them around, the sequential order in the INBOX will get thrown off. So the phone’s light will stay on, because Asterisk can see a file(s) in there, but when they go to access their voicemail, it’ll say they have no messages, because the voicemail system doesn’t see a msg0.wav file, instead there would be a msg6.wav file or something like that in there. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. SilerSent: Monday, November 08, 2004 2:20 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] MWI Doesn't Turn Off Anyone having issues with the message indicator lights after CVS-HEAD-07/23/04-13:55:59 ?? For several of my users, our MWI lights do not turn off. Phones are Polycom IP500 and this just started prior to my last update. Should I update to a newer version? I pulled this from the CVS last week so I thought it was newest. Thanks, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI Doesn't Turn Off
Anyone having issues with the message indicator lights after CVS-HEAD-07/23/04-13:55:59 ?? For several of my users, our MWI lights do not turn off. Phones are Polycom IP500 and this just started prior to my last update. Should I update to a newer version? I pulled this from the CVS last week so I thought it was newest. Thanks, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Running Asterisk on Linksys Router
Any documentation on how to do this anywhere? W From: Brian C. Fertig [mailto:[EMAIL PROTECTED] Sent: Thursday, October 14, 2004 12:52 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Running Asterisk on Linksys Router ahh my bad.. Didn’t know you could run on that.. I will have to look into it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of christophe de coninckSent: Thursday, October 14, 2004 3:39 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Running Asterisk on Linksys Router I think it would be running on a linksys wrt54g since that are the ones were you are capable to put your own linuxdistro on it and run your own tools on it like example iptables and asterisk.That's what he ment I think, not putting it in the DMZ.On Thu, 2004-10-14 at 21:35, Brian C. Fertig wrote: I run asterisk at my house on a linksys router. I have it sitting inthe DMZ of the router so it acts like its outside. Works perfectlyfine. .o---o.Brian FertigNetwork EngineerPlanet Telecom, Inc.Tampa, FL Office813.864.3161x107 Office813.864.3164 Direct813.817.9961 Cellular813.881.9762 FaxWeb: www.planet-telecom.comemail: [EMAIL PROTECTED]-->IM's<---MSN: [EMAIL PROTECTED]AIM: ptelebrianYahoo: ptele_brian -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of James H.ThompsonSent: Thursday, October 14, 2004 3:17 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Running Asterisk on Linksys Router At Astricon Mark mentioned that somone had Asterisk running on a LinksysRouter.Anyone have more information on this? Jim James H. Thompson[EMAIL PROTECTED] ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christophe De Coninck | Zarek K http://www.zarekk.bemailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] musiconhold will not start
99.9% sure not sound card is required for MOH. I don't think you want the latest version of MPG123. Think you want mpg123 0.59r only not s-r4 Make sure to copy all mp3 files (if over FTP) using the binary transfer method only. W -Original Message- From: Andy Reinke [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 12, 2004 3:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] musiconhold will not start I assume that the problem with /dev/dsp is your issue - haven't confirmed but I bet * uses the soundcard for music on hold. I often see that message when I am logged into an X session - the first desktop on the console gets the sound. Be sure to log out and restart *. Then reconfigure X desktop applets so you don't load any sound applets or programs on login. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Tuesday, October 12, 2004 5:10 PM To: Asterisk Users List Subject: [Asterisk-Users] musiconhold will not start I have * running on gentoo. Everything seems to be working fine but musiconhold will not start. When starting * I get these errors, but guess that's not the problem: Oct 12 16:42:12 WARNING[16384]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled Oct 12 16:42:12 WARNING[16384]: chan_oss.c:434 soundcard_init: Unable to open /dev/dsp: No such file or directory When MOH needs to kick in however I get this message: WARNING[294927]: res_musiconhold.c:366 moh1_exec: Unable to start music on hold (class '30') on channel SIP/101-8168 The box has media-sound/mpg123 Latest version installed: 0.59s-r4 I'm not sure why it cannot start the muzak however, the wiki says that a symlink must be created to the binary but the binary is already in place where the symlink should come. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Echo
I have Polycoms and I sometimes have echo. However, it is always on an incoming call and always a matter of echo training. Have you already worked through all the settings for ech in *? Are you adjusting rx and tx gains? Do you have echo cancellation set to high? What are the rest of the parameters for your particular problem? W -Original Message- From: Matthew Marlowe [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 12, 2004 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom Echo Lately I have been experiencing a lot of echo from my Polycom phones. Only I hear the echo and it's not on every call. I've researched it via google and the forums and every echo problem usually relates when it's using a Zap card and not an IAX provider. Can anyone give me some advice or where to look to help solve this echo problem? This never occurs on any of our other phones, Ciscos, Grandstreams, Sipuras, etc.. Only on the polycoms. Any help would be greatly appreciated. Thanks in advance. -- MBM ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Alternate MP3 Player
Search here... http://www.voip-info.org There are alternatives... -Original Message- From: Leah Newmark [mailto:[EMAIL PROTECTED] Sent: Thursday, September 23, 2004 1:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Alternate MP3 Player Hi! I am currently working on setting up an Asterisk system, and I was wondering if anyone has worked on an alternate mp3 player to mpg123. We have a library of MP3 files that we would like people to be able to select and play over the phone -- and this will require pause & resume, as well as fast forward / reverse (jump forward / jump back). It doesn't seem like mpg123 can do this. Is there any application that can, that is also compatible with Asterisk? Thank you for all your help! Leah Newmark Capalon Hosting Solutions ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] new user From Canada
Go here.. http://www.voip-info.org search on IVR Everything is driven out of extensions.conf Regards, Wiley From: Jose J. Avalis [mailto:[EMAIL PROTECTED] Sent: Thursday, September 23, 2004 12:42 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] new user >From Canada Hi all, We are currently using an IVR system, we are Java developers and this solution seems to be cool. Can anyone tell me where can I find more details about the programming of this IVR, specially ( call flows ) , I’m looking for some programming tool or at least specs and commands. Thanks The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone
The free phones I have heard of are soft phones... X-Lite is excellent... Cheap phones to test with, Grandstream is cheap but like the man said, you get what you pay for. eBay is a good source for cheap phones to test with but cheap is relative I consider cheap as sub $100. You can pick up a couple of models of Cisco and Polycom for 100-150. W -Original Message- From: Huddleston, Robert [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 8:04 AM To: 'Shaun Ewing'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone Anyone know where we could get a cheap sip phone... We've been playing with an Innomedia MGCP and SIP adapters and failing - so thinking that testing with a real phone might be good.. Robert A. Huddleston, KF4BYY IT Support Analyst Cavalier Telephone LLC. (Cell) 804.400.3686 [EMAIL PROTECTED] -Original Message- From: Shaun Ewing [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 11:04 AM To: Michael Bielicki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone On Wed, 22 Sep 2004 16:40:04 +0200, Michael Bielicki <[EMAIL PROTECTED]> wrote: > Cisco 7940 :) I'll concur with that. The Cisco 7940 and 7960 phones have great speakerphones :) As for ones to stay away from - the Grandstream BT-100 series. The sound is fine on the local end, but is very low for the remote end (sounds as if the microphone in speaker mode is actually the mic on the handset). -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone
Do you have a price range? I use Polycom IP500s and the speaker phone is awesome. It picks up speakers in the room very well at 5-6 feet. Polycom has always made an exceptional speaker phone even on plain ole phones. Their implementation on the IP phones is excellent so they are my preference. I have heard that the Cisco phones are quite nice too. I think from a previous conver that the 7905 has a speaker phone and is priced fairly low. Cheers, Wiley -Original Message- From: Michael Bielicki [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Cisco 7940 :) - Original Message - From: Phil Siegrist <[EMAIL PROTECTED]> Date: Wed, 22 Sep 2004 10:15:57 -0400 Subject: [Asterisk-Users] SIP Phone To: [EMAIL PROTECTED] Hi All, I am look for recommendations for a good SIP phone, specifically with a good speaker phone. I have tried the SNOM 100 and the speaker phone quality is quite poor. Can any one share there experiences with this. Much Appreciated, Phil ___ 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 -- Michael Bielicki ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Cisco 7905G
No worries. You are totally right that $8 bucks is pretty neglibable for that service. Your comment about the vendor made me think of a fellow I recently helped with his * who resells Cisco. His company is called TekSavers and they have the phone that started this email (Cisco 7905G) for $95. I haven't a clue if that is a good price but your comment made me think of his site. http://www.teksavers.com I don't know if that will be of use but I just thought I would pass it on in case there are any items out there that are priced good. Cheers, Wiley -Original Message- From: Michael Loftis [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7905G --On Tuesday, September 21, 2004 15:58 -0700 "Wiley E. Siler" <[EMAIL PROTECTED]> wrote: > A completely valid point you make... > > Just remember to multiply that 8 dollars times the number of phones > times the number of years you need to have them in service > Extend that to the other Cisco items you need to maintain and purchase > and the costs keeps rising > Mine is just a simple observation regarding the costs of being a Cisco > customer. > > Like I said, it is about preference and budgeting... > Even if I only save $300 dollars a year from using alternatives to > Cisco in my enterpise (switches, firewalls, phones, etc), it is still $300. > Usually, the $ saved from finding comparable products at lower prices > also benefit my overall budget since Cisco tends to be a premium price. > > Like I said... I won't disparage Cisco products in any way. They are > barring none, some of the best out there. > I just prefer a different licensing model for my enterprise and find > that I can get comparable performance by alternatives. > If anyone has the budget and desire, they would be well served by any > Cisco product they purchase. > > It is just not my choice... Thus my disclaimer in the footer of the > last post. Ohhno, I'm not grilling you or anything Wiley, just pointing out that for many orgs the $8/mo wouldn't be too bad considering the level of support they'd receive. For some it's not worth it, and that's totally understandable. We each have our own priorities to consider. I simply wanted to mention the actual cost involved in case anyone thought that it was some really absurd per-phone price. My biggest problem still remains finding competent resellers where I can buy the phones from. > > Cheers, > Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Cisco 7905G
A completely valid point you make... Just remember to multiply that 8 dollars times the number of phones times the number of years you need to have them in service Extend that to the other Cisco items you need to maintain and purchase and the costs keeps rising Mine is just a simple observation regarding the costs of being a Cisco customer. Like I said, it is about preference and budgeting... Even if I only save $300 dollars a year from using alternatives to Cisco in my enterpise (switches, firewalls, phones, etc), it is still $300. Usually, the $ saved from finding comparable products at lower prices also benefit my overall budget since Cisco tends to be a premium price. Like I said... I won't disparage Cisco products in any way. They are barring none, some of the best out there. I just prefer a different licensing model for my enterprise and find that I can get comparable performance by alternatives. If anyone has the budget and desire, they would be well served by any Cisco product they purchase. It is just not my choice... Thus my disclaimer in the footer of the last post. Cheers, Wiley -Original Message- From: Michael Loftis [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7905G I'm pretty sure SMARTnet on the VoIP phones is like $8/phone/yearMost orgs spend more than that per person on electricity for CRTs. --On Tuesday, September 21, 2004 15:25 -0700 "Wiley E. Siler" <[EMAIL PROTECTED]> wrote: > See my post of a few moments ago and you have hit on the exact reason > I will not use Cisco beyond my firewall (a purchase you will never > regret if you need a good firewall). Cisco makes arguably some of (if > not > totally) the best equipment out there. I just have one problem. > > Their licensing model is such that you can buy their product, at a > premium price too mind you, ant then you have to pay MUCH extra for a > support contract just to get images and just about everything else you > need. That lead me to alternatives. Polycom, SNOM, Grandstream are > just a few and each good based upon certain criteria (price vs. looks > vs. performance, etc, etc). > > My personal choice is Polycom. Polycom IP300 phones are excellent if > you do not need speaker phone. IP500 is excellent with all the > features you may want. Go to the IP600 and you get a minibrowser > though the benefit is arguable. My IP500s perform extremely well and > the featureset is excellent. > > As before, phone choice is very preferential and what I like may be > totally hated by someone else. However, in my opinion, the > dollar/performance/presentation ration of these phones is excellent. > Not to mention that the SIP images are available online as released > instead of regulated the same way Cisco is. I just cannot bring > myself to pay the Cisco premium for hardware then have to give them > even more money for the things I need to make THEIR hardware work > right. That would just seems like Cisco is sticking it to me too much... > > $0.02 > > Wiley > > PS. I have done my best to express that this is MY preference so be > sure to weight all the opinions you find. Many have extremely good > results using the Cisco phones and can justify the cost thusly. My > enterprise just isn't built on that large a budget. > > > > > > > -Original Message- > From: Gunnar Andersson [mailto:[EMAIL PROTECTED] > Sent: Tuesday, September 21, 2004 2:58 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Cisco 7905G > > Hi All > > Just received my first 7905G from a distributer here in Sweden. > According to the spec this phone should be able to use SIP. Now I been > looking on Ciscos home pages for several hours trying to find a "SIP > image" for this phone. > No luck at all, need special access to be able to download software to > this phone. Is it the fact, that I have to pay for a contract of some > kind to be able to use this phone with SIP and *. > This was the first product we bought from Cisco... and maybe the last. > > rgds > > Gunnar Andersson > > ___ > 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 > > > The information transmitted is intended only for the person or entity > to which it is addressed and may contain confidential and/or > privileged material. Any review, retransmission, dissemination or > other use of, or taking of any action in reliance upon, this > informatio
RE: [Asterisk-Users] Cisco 7905G
See my post of a few moments ago and you have hit on the exact reason I will not use Cisco beyond my firewall (a purchase you will never regret if you need a good firewall). Cisco makes arguably some of (if not totally) the best equipment out there. I just have one problem. Their licensing model is such that you can buy their product, at a premium price too mind you, ant then you have to pay MUCH extra for a support contract just to get images and just about everything else you need. That lead me to alternatives. Polycom, SNOM, Grandstream are just a few and each good based upon certain criteria (price vs. looks vs. performance, etc, etc). My personal choice is Polycom. Polycom IP300 phones are excellent if you do not need speaker phone. IP500 is excellent with all the features you may want. Go to the IP600 and you get a minibrowser though the benefit is arguable. My IP500s perform extremely well and the featureset is excellent. As before, phone choice is very preferential and what I like may be totally hated by someone else. However, in my opinion, the dollar/performance/presentation ration of these phones is excellent. Not to mention that the SIP images are available online as released instead of regulated the same way Cisco is. I just cannot bring myself to pay the Cisco premium for hardware then have to give them even more money for the things I need to make THEIR hardware work right. That would just seems like Cisco is sticking it to me too much... $0.02 Wiley PS. I have done my best to express that this is MY preference so be sure to weight all the opinions you find. Many have extremely good results using the Cisco phones and can justify the cost thusly. My enterprise just isn't built on that large a budget. -Original Message- From: Gunnar Andersson [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 2:58 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7905G Hi All Just received my first 7905G from a distributer here in Sweden. According to the spec this phone should be able to use SIP. Now I been looking on Ciscos home pages for several hours trying to find a "SIP image" for this phone. No luck at all, need special access to be able to download software to this phone. Is it the fact, that I have to pay for a contract of some kind to be able to use this phone with SIP and *. This was the first product we bought from Cisco... and maybe the last. rgds Gunnar Andersson ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Asterisk , ISA Firewall/VPN , STUN and other issues
I use my XLite softphone from my Win XP box over VPN to my Cisco PIX with no issues so this can be done. How that works for an ISA box is unknown to me. I dumped ISA several years ago do to it's (IMHO) unpredictability and low performance. Are you using the built in VPN of WinXP or an ISA Client ? Phones are very preferential. Grandstream and SNOM make some good cheap phones if presentation is not an issue. I personally prefer Polycom IP300 and IP500 for ease of use and features. IP300 can be had for $135 on eBay. Some like Cisco though the Cisco licensing model irritates me to no end and I refuse to use them for anything other than firewall/router at this time. Cheers, Wiley From: Shawn Dillon [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 2:39 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk , ISA Firewall/VPN , STUN and other issues I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone , on the SIP client nothing. The Debian machine has two NIC’s , one with a static external IP and one with an internal IP. Our remote offices are behind a mixture of firewalls. I have some questions with regards to our testing and setup. 1) Is there a way to get the SIP/IAX client to work via the VPN? This would be the easiest way. 2) If not can I install a STUN server on the same machine as the * server? Can it use the same internal and external IP’s as the * server? 3) Is there a hardphone that supports VPN that has been tested? 4) What is the best hardphone to use with Asterisk? Thanks for the input Shawn Dillon The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Auto Attendant How To ?
Here is the best starting point. It is all driven out of extentions.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20ivr%20menu Search for IVR and you will find good info... Cheers, W From: Luis Czop [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 12:13 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Auto Attendant How To ? Hi friends, Does anyone know where can I find an "How To" to config the auto attendant? Anything else? Many thanks in advance Luis Eduardo Czop Gte. de Tecnología y Servicios PMS Argentina SA Av. Alicia M. de Justo 170 - Piso 1° (C1107AAD) Ciudad de Buenos Aires Tel: 5217 9311 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Music on hold
Did you transfer the mp3s to the * box via FTP? If so, did you use binary not ascii mode? Ascii mode will mess the files up every time. Explicitly call binary then your mget so your files don't get hosed. Also, if your musiconhold.conf file has a good reference to where your mp3s are located... default => quietmp3:/var/lib/asterisk/mohmp3,-Z Then are you calling the context from your extensions.conf file correctly? I use this extension just so I can test that the music is there... ; ;MUSIC ON HOLD EXTENSION ; exten => 6000,1,Answer exten => 6000,2,SetMusicOnHold(default) exten => 6000,3,MusicOnHold() exten => 6000,4,Hangup Dialing from my SIP phone produces tunes as expected and can of course also be verified afterwards via in incoming call put on hold... Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, September 20, 2004 4:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold > Hello List! > > I followed the instruction from > http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf to get > my music working when i put someone on hold. I have tried it with this mpg123 version as well: --- High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Still silence, but a mpg123 process is running. Used codec is ulaw. Thanks, Mario ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] 1 extension entry for multiple purposes?
My solution assumes you are capturing the caller ID of internal users
and redirecting them to the password prompt to save time and effort.
All internal call recipients should see the correct info (name and
extension). For an external call, you may need to set that info for
your BRI/PRI. Whatever you are using... For me, it is set by the phone
company since I am using POTS lines. If you are using BRI or PRI, there
is info on the wiki about it and how you can send the correct caller ID
to an external recipient.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerI
D
This gave a lot of results at google: site:lists.digium.com PRI
CallerID
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Monday, September 20, 2004 3:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1 extension entry for multiple purposes?
pbx*CLI> show dialplan cytel-outgoing
[ Context 'cytel-outgoing' created by 'pbx_config' ]
's' =>1. SetCIDName(${CALLERIDNAME})
[pbx_config]
2. SetCIDNum(2814494000) [pbx_config]
'_9011.' => 1. Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[pbx_config]
'_91XX' => 1. Dial(SIP/${EXTEN:[EMAIL PROTECTED]) [pbx_config]
'_9281XXX' => 1. Dial(SIP/${EXTEN:[EMAIL PROTECTED]) [pbx_config]
'_9713XXX' => 1. Dial(SIP/${EXTEN:[EMAIL PROTECTED]) [pbx_config]
'_9832XXX' => 1. Dial(SIP/${EXTEN:[EMAIL PROTECTED]) [pbx_config]
Shouldn't that 's' context apply those 2 priorities first then find the
pattern below?
Matthew
- Original Message -
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, September 20, 2004 5:14 PM
Subject: Re: [Asterisk-Users] 1 extension entry for multiple purposes?
> I never said it didn't work. I'm saying that if I use callerid= in the
> sip.conf in conjunction with the 1-line-voicemailmail that Wiley
showed
me,
> it won't work.
>
> Since Wiley's fix was to use ${CALLERIDNUM} in the voicemailmain
exten, if
I
> have callerid=999-999- in sip.conf then VoicemailMain will use
> 99 instead of the extension the person is calling from.
>
> OK. So I removed all the callerid= from the sip.conf and Wiley's fix
works
> perefectly. But I am back to where if I call out, the caller id shows
up
as
> my extension only.
>
> My fix, that didn't work:
>
> [global-outgoing]
> exten => s,1,SetCIDNum(212-433-3344)
> exten => _9212XXX,2,Dial(SIP/${EXTEN}@,15,tr)
> exten => _91XX,2,Dial(SIP/${EXTEN}@,15,tr))
>
> I figured that if I tacked an 's' extension before the pattern
matching,
> every outgoing pattern below the 's' would get that CID. But that
didn't
> work.
>
> Matthew
> - Original Message -
> From: "Marc Storck" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Monday, September 20, 2004 4:57 PM
> Subject: Re: [Asterisk-Users] 1 extension entry for multiple purposes?
>
>
> > this works great for me, i use callerid= like this:
> >
> > callerid="Marc Storck" <35227273033>
> >
> > Matthew Boehm wrote:
> >
> > > OK. Here is the caveat I've found. The phones, in sip.conf, all
have a
> > > callerid= line because if they don't when they call someone the
caller
> id
> > > shows up ONLY as their extension.
> > >
> > > For instance, my extension is 3044. When I call my cell, all it
says
is
> > > "Missed call from 3044".
> > >
> > > The only way I found to fix this was to add that callerid= into
the
> sip.conf
> > >
> > > But since I have done that, what you have suggested below won't
work.
> > >
> > > Should I have the callerid set somewhere else?
> > >
> > > Matthew
> > > - Original Message -
> > > From: "Wiley E. Siler" <[EMAIL PROTECTED]>
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > <[EMAIL PROTECTED]>
> > > Sent: Monday, September 20, 2004 4:14 PM
> > > Subject: RE: [Asterisk-Users] 1 extension entry for multiple
purposes?
> > >
> > >
> > > Here you go... No extension required
> > >
> > >>From extensions.conf
> > >
> > > ;--
> > > ; VOICEMAIL ENTRY INTO SYSTEM
> > >
RE: [Asterisk-Users] 1 extension entry for multiple purposes?
Here you go... No extension required
>From extensions.conf
;--
; VOICEMAIL ENTRY INTO SYSTEM
;--
exten => 8,1,Answer
exten => 8,2,Wait(1)
exten => 8,3,VoicemailMain(${CALLERIDNUM})
exten => 8,4,Hangup
Still want the old way of enter your number then PIN...
exten => 81,1,VoicemailMain2()
exten => 81,2,Hangup
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Monday, September 20, 2004 2:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 1 extension entry for multiple purposes?
Hey gang,
There must be any easy solution for this but my mind is frazzled on
compiling 2.4 with RTC as module. Bleh.
Currently extension 9000 is our VoicemailMain(@company) line. Some
employee's are complaining that the old system was better because you
didn't have to enter your mailbox number and that instead the old system
took you right to it.
I figured there was something similar so that I don't have to have 200
extra extensions.conf lines just for VoicemailMain(@company).
Basically I want something like this: exten =>
9000,1,VoicemailMain([EMAIL PROTECTED])
so that way all it asks for is their password.
Any ideas..?
Thanks,
Matthew
___
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RE: [Asterisk-Users] Newbie has a few basic questions please.
Bruce, Using a POTS line local with * will get you the same net result as having the POTS line only. You will be using VoIP internal and passing your calls off to the * box to have it dial like a normal phone. So, no IP packets move past the box over a POTS line. That is a pretty useful feature for someone in my situation where I needed to get a PBX in place using available infrastructure (we had the POTS and didn't want to pay for an upgrade to a PRI T1). I just connected up my lines and use my * like a PBX. So, if you want to use VoIP exclusively, you would need to pass all your traffic over an internet connection. In theory, I think you can use a digium card with an fxs to connect up an analog phone to the * box. Then you would just route the call out of your data connection to someone like IAXtel (http://www.iaxtel.com) where you have registered yourself and your parent's * box. The call would pass to their VoIP connection and handoff to whatever their * box had connected (analog phones, SIP, whatever). Peruse the Digium website FAQ and you should come up with lots of different info regarding what you can do. Thanks, W -Original Message- From: Bruce [mailto:[EMAIL PROTECTED] Sent: Monday, September 20, 2004 12:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie has a few basic questions please. I think I am missing the whole purposes of *. i see that it can do mainy things, but in laymans temrs I am not sure what it does. I am very proficient in Linux and would like to use * for the following: 1) I would like to get rid of my landline(verizon) and use voip as my main means to communicate on the telephone. I would like to be able to plug in my plain old phone into my linux box and be able to make a phone call to my family who has a plain old telephone line going into thier house, using voip and then I guess connecting to the pstn. Can i do this? If so, how? What hardware do I need? Can anyone connect to PSTN lines for free? Or do I need to pay some phone company somewhere? thanks to anyone who can help. Bruce ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] Cheap Sams computer good for tdm400?
Felix, You might try going to Sam's (or their website) and looking for the motherboard manufacturer on their marketing materials. Then you can get the specs for the motherboard from the mobo maker. I cannot imagine anyone here will know if the PC you are reference is compliant of the tops of their heads (who knows, i could be wrong). I bet most of them would look that info up from hitting the Sams site as suggested. However, you should be able to find this information too if you go to the sams website. For information regarding scaling of Asterisk, you can query this user list by going to google and typing: site:lists.digium.com While a cheapo box will work, it definitely won't scale to a large amount of users. Cheers, Wiley From: Felix Pizarro [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 14, 2004 5:24 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Cheap Sams computer good for tdm400? I need a cheap platform for installing a tdm400. Could someone tell me if the cheap cpubuilders computer at sams $179 (cbs110l) is pci 2.2 compliance? I ve got a compaq deskpro en 700 that does not seems to be compliant and I need to change it to start developing. Thanks for the help. Computer Model: CBS110L Do you Yahoo!?New and Improved Yahoo! Mail - Send 10MB messages! The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] music on hold not strting
I never stripped my tags and things work fine for me. I had problems at first too with MOH. My problem was due to how I was copying over the files. I was copy via FTP using the command line in Linux. However, if you do not explicitly state binary as the copy method, it will copy the files over using ASCII. Doing so mungs the whole MOH player and never worked right. Issueing the binary command at the ftp command prompt prior to pulling the files to my asterisk box solved the problem for me. I do not know your method of copying to your server but this may be your problem if you are using FTP. Cheers, Wiley -Original Message- From: Andreas Roedl [mailto:[EMAIL PROTECTED] Sent: Monday, September 13, 2004 6:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] music on hold not strting Hello! Am Montag, 13. September 2004 14:40 schrieb Altus Snyman: > In the howto it tells me I should strip the ID3 tags How do I do that? http://fibiger.org/mp3tag.html Andi ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP on Handhelds
What was the symptom of the sound problem?
Echo cancellation on the * box should make the call sound good for the standard * user
side I would think. Does the sound quality suffer on the iPaq side then?
I wonder if the encoding process on the iPaq is to blame for bad sound at the * side.
What model of iPaq was used?
Thanks,
Wiley
-Original Message-
From: Bill Seddon [mailto:[EMAIL PROTECTED]
Sent: Sat 9/11/2004 1:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [Asterisk-Users] SIP on Handhelds
We've installed and used PPC X-Lite on an iPAQ with 802.11b. While the
sound quality of the iPAQ user was OK (not great but OK) the sound quality
as heard by the other caller was very poor.
If you try using a softphone on a PPC, I'd be interested to hear of your
experience.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler
Sent: September 11, 2004 12:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP on Handhelds
Thank you! Found the link here...
http://www.freewareppc.com/communication/xlite.shtml
-Original Message-
From: Iassen Hristov [mailto:[EMAIL PROTECTED]
Sent: Friday, September 10, 2004 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP on Handhelds
As far as I can tell, both SJphone and X-Lite offer PocketPC versions.
--On Friday, September 10, 2004 4:25 PM -0700 "Wiley E. Siler"
<[EMAIL PROTECTED]> wrote:
> I think the Bluetooth requirement may be where that hangs up. I want
> to be able to setup an handsfree headset too.
>
> I am thinking I will either write a sip based client in .net using the
> RTC API or implement the IAX model you reference here.
>
> Thank you!
> Wiley
>
>
>
>
> -Original Message-
> From: Matt Gibson [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 10, 2004 4:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP on Handhelds
>
> If you're not opposed to running linux the sharp zaurus 6100 has
> 802.11b built in
>
> http://www.sharpusa.com/products/TypeLanding/0,1056,112,00.html
>
> and there's a client available to connect to iax on voip-info.org. I
> know you asked for SIP, but.. this is all that's avail i can find :)
>
> http://www.kauss.org/Stephan/ziaxphone/
>
>
> matt
>
>
> Wiley E. Siler wrote:
>
>> Does anyone know if SIP will/is support on handheld PCs such as the
>> iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems
>> like a solution to provide a wireless based sip phone for any user
>> would be possible. Handoff between access points might be
>> problematic
>
>> but most users I know would be using their PDA phone in an airport
>> with free wireless or at the local cafe, etc, etc...
>>
>> Can anyone with experience in this department let me know if they
>> think this idea is possible?
>>
>> Thanks,
>> Wiley
>>
>>
>>
>>
>>
>>
>> -
>> -
>> --
>>
>> The information transmitted is intended only for the person or entity
>> to which it is addressed and may contain confidential and/or
>> privileged material. Any review, retransmission, dissemination or
>> other use of, or taking of any action in reliance upon, this
>> information by persons or entities other than the intended recipient
>> is prohibited. If you received this in error, please contact the
>> sender and delete the material from any computer
>>
>> -
RE: [Asterisk-Users] SIP on Handhelds
Thank you! Found the link here... http://www.freewareppc.com/communication/xlite.shtml -Original Message- From: Iassen Hristov [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP on Handhelds As far as I can tell, both SJphone and X-Lite offer PocketPC versions. --On Friday, September 10, 2004 4:25 PM -0700 "Wiley E. Siler" <[EMAIL PROTECTED]> wrote: > I think the Bluetooth requirement may be where that hangs up. I want > to be able to setup an handsfree headset too. > > I am thinking I will either write a sip based client in .net using the > RTC API or implement the IAX model you reference here. > > Thank you! > Wiley > > > > > -Original Message- > From: Matt Gibson [mailto:[EMAIL PROTECTED] > Sent: Friday, September 10, 2004 4:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP on Handhelds > > If you're not opposed to running linux the sharp zaurus 6100 has > 802.11b built in > > http://www.sharpusa.com/products/TypeLanding/0,1056,112,00.html > > and there's a client available to connect to iax on voip-info.org. I > know you asked for SIP, but.. this is all that's avail i can find :) > > http://www.kauss.org/Stephan/ziaxphone/ > > > matt > > > Wiley E. Siler wrote: > >> Does anyone know if SIP will/is support on handheld PCs such as the >> iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems >> like a solution to provide a wireless based sip phone for any user >> would be possible. Handoff between access points might be >> problematic > >> but most users I know would be using their PDA phone in an airport >> with free wireless or at the local cafe, etc, etc... >> >> Can anyone with experience in this department let me know if they >> think this idea is possible? >> >> Thanks, >> Wiley >> >> >> >> >> >> >> - >> - >> -- >> >> The information transmitted is intended only for the person or entity >> to which it is addressed and may contain confidential and/or >> privileged material. Any review, retransmission, dissemination or >> other use of, or taking of any action in reliance upon, this >> information by persons or entities other than the intended recipient >> is prohibited. If you received this in error, please contact the >> sender and delete the material from any computer >> >> - >> -- >> - >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP on Handhelds
I think the Bluetooth requirement may be where that hangs up. I want to be able to setup an handsfree headset too. I am thinking I will either write a sip based client in .net using the RTC API or implement the IAX model you reference here. Thank you! Wiley -Original Message- From: Matt Gibson [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP on Handhelds If you're not opposed to running linux the sharp zaurus 6100 has 802.11b built in http://www.sharpusa.com/products/TypeLanding/0,1056,112,00.html and there's a client available to connect to iax on voip-info.org. I know you asked for SIP, but.. this is all that's avail i can find :) http://www.kauss.org/Stephan/ziaxphone/ matt Wiley E. Siler wrote: > Does anyone know if SIP will/is support on handheld PCs such as the > iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems > like a solution to provide a wireless based sip phone for any user > would be possible. Handoff between access points might be problematic > but most users I know would be using their PDA phone in an airport > with free wireless or at the local cafe, etc, etc... > > Can anyone with experience in this department let me know if they > think this idea is possible? > > Thanks, > Wiley > > > > > > > -- > -- > > The information transmitted is intended only for the person or entity > to which it is addressed and may contain confidential and/or > privileged material. Any review, retransmission, dissemination or > other use of, or taking of any action in reliance upon, this > information by persons or entities other than the intended recipient > is prohibited. If you received this in error, please contact the > sender and delete the material from any computer > >--- >- > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP on Handhelds
Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can anyone with experience in this department let me know if they think this idea is possible? Thanks, Wiley The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser
Yep. Those are the latest. Sip 1.3 and Boot 2.5. Thank you! Wiley > -Original Message- > From: Patrick [mailto:[EMAIL PROTECTED] > Sent: Monday, August 16, 2004 4:11 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 500/600 > XML minibrowser > > On Tue, 2004-08-17 at 00:34, Wiley E. Siler wrote: > > Also, is the new SIP and bootrom release available for download > > somewhere? > > > > Thanks, > > Wiley > > Don't know if these are the latest but here are some links. > First one has sip & bootrom files: > http://www.freedomphones.net/polycom/files/ > http://www.voip-info.org/wiki-Polycom+Phones > > Regards, > Patrick > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soft DSS for Asterisk
Is there a Software based DSS application available for Asterisk? Thanks, Wiley Siler
RE: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser
Also, is the new SIP and bootrom release available for download somewhere? Thanks, Wiley > -Original Message- > From: Derek Listmail Acct [mailto:[EMAIL PROTECTED] > Sent: Monday, August 16, 2004 2:39 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML > minibrowser > > Has anyone been able to get the minibrowser on the Polycom > SoundPoint IP 500/600 phones working? If so could you share > the relevant sections of your config with me? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ 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] 123 Basic configuration files
Best starter examples http://www.automated.it/guidetoasterisk.htm Documentation http://www.digium.com/index.php?menu=documentation Asterisk will make sample files for you... read teh doucmentation at the first link I listed... Regards, Wiley I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk [EMAIL PROTECTED] www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home
RE: [Asterisk-Users] Free MOH MP3
Well, yes it is. Sorry about that. I didn't even think about the Wiki since what I was looking for was content. I just googled against the list thinking that was where I saw it. Thanks! Cheers, Wiley > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Saturday, August 14, 2004 5:01 PM > To: 'Bill Church'; [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Free MOH MP3 > > The wiki is your friend, found it in under 30 seconds. > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHol d > > Under also see: > > * Sounddogs http://www.sounddogs.com/catsearch.asp?Type=2 Royalty Free > Music > * FreeMusic http://hebb.mit.edu/FreeMusic/ Free Classical Music > > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler > Sent: Saturday, August 14, 2004 7:51 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Free MOH MP3 > > > > Hello All, > > > > Sorry to rehash a question I am sure has shown several time but I cannot > google up the answer from the lists. > > > > Does anyone know where I can get some royalty free, cost free music for my > music on hold? > > > > I saw someone's post several weeks ago that said that this exists at a > download site but I have not been able to find it. > > > > Thanks! > > Wiley Siler > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free MOH MP3
Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone’s post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler
RE: [Asterisk-Users] Help - is voip good for in-house calls?
Hello Francis, > I'll most likely use a BRI. Do you think this will help to avoid echo? I could not say as I have never used a BRI and I am pretty new to this too. I do know that BRI is supported from watching conversations in this email list and reading online. People seem to use it a bit so it must work well. Googling the list with BRI should get you tons of good leads. Greg had a great idea in having you set it up and try it. In fact, that is exactly how I did mine. I purchase a cheap clone card for $15 and used it to test on one POTS line while I tweaked my configuration files and got the system validated. I tested the system with soft phones, one Polycom IP 500, and one Grandstream Budgetone 101. The Budgetone worked well and was leagues easier to setup than my Polycom actually. For expandability, I believe that the cap I have seen is about 60 concurrent calls for one Asterisk box and that is with a pretty serious server by most users standards. I cannot imagine having that many calls at this point so I am fine but I jus though t you would want to know. The nice thing about * is that you can just build another server and link them together over IAX. Again, the low cost of implementation pays off and you get to continue growth. I will never go back to proprietary PBX now that I finally have a solution that I can control. Cheers, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help - is voip good for in-house calls?
Hello Francis, My office build is the same as yours. 15 or so extensions, low traffic 100MB network, and a desire to have a phone system that uses VoIP. I have my system working as a PBX just like you would. I use two TDM400s for my 8 POTS lines and Polycom IP 500 phones at the desktop. I also tested with the Grandstream phones you suggested. SO, we have the same system requirements so here are the answers as I have found them for my implementation Voice quality on the SIP based phones has a lot to do with the codec you use. The lowest compression codec is uLaw and that is what I use since we have tons of bandwidth to spare. Also, my HP switch has COS (class of service which is like QOS) so I can prioritize the packets coming from my phones over the standard network traffic. Even without this switching feature turned on, performance was great. The phones themselves play another role in the quality. Grandstreams are pretty good and I have only used mine for testing so I will not disparage them. However, the old saying stands. You get what you pay for. Raising your phone budget from $85 to more like $150-250 will get you a phone with more features and greater expandability in my IHO. However, you can still do great things with the cheaper Grandstream phones and still have a system that works very well. IT is all up to what you can spend and what you need. Google the archive by putting "site:lists.digium.com" in front of your search string (no quotes though). You should see some good info on phones. Latency is gonna be there on any network. However, on my network (which is just like yours) the latency is very very low. We are talking 20-40ms tops and it is completely unnoticeable when using the phone. The only problem I have had at all has been with occasional echo. It does not happen often and it usually takes about 5 seconds for the * box to train up and remove it. Most of this seems to originate in the fact that I am using POTS lines. The solution that uses a T1 PRI has better features and I think it has less echo potential. However, that would not work for me since my T1 provider wanted to make me pay 6 grand to switch to a PRI from my standard data T1 with POTS. Just some food for thought... I have been a VoIP user for about 1 month after spending another researching what when where how... So, we know I am not an expert... but as a fellow user and new VoIP initiate, I can tell you that Asterisk is a phenomenal product for SMB level offices like yours and mine. When I compared it to a PBX system of comparable power, expandability, and feature set, Asterisk won easily since the only real cost I have had was for my phones. I have my system in place for around 3000 dollars and it is competitive with all the 10K dollar solutions the vendors threw at me plus it has an undeniable advantage in upgrade path. All upgrades to the system are free and the sky is the limit to what you can build using the framework that all the * gurus have built into this system. Not to mention the fact that if anything ever goes wrong with the server, I can have a new one in place in under and hour. Try that with a PBX when some proprietary part goes belly up. You could wait days potentially. My $.02. Hope this helps. Cheers, Wiley -Original Message- From: Francis Augusto Medeiros [mailto:[EMAIL PROTECTED] Sent: Saturday, August 14, 2004 1:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help - is voip good for in-house calls? Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and standards as a PBX-only, with "traditional" phones? Or should I go all the way to Digium's TDM? Or should I forget the whole thing and get a traditional PBX? ;) My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. I've read lots about voip, and I'm quite impressed with it, but most case studies show voip being used to interconnect offices. My case is different - I want to replace a traditional PBX to handle in-house phone calls, either from room-to-room in the same building and room-to-POTS. Any comment, help, tip or link would be greatly appreciated! Yours truly, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options v
RE: [Asterisk-Users] Polycom IP 500 - MWI Not Working
Nope. That fixed it. Thank you! Wiley -Original Message- From: Robert Jackson [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 10, 2004 2:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 - MWI Not Working >-Original Message- >From: Wiley E. Siler [mailto:[EMAIL PROTECTED] >Sent: Tuesday, August 10, 2004 4:33 PM >To: [EMAIL PROTECTED] >Subject: [Asterisk-Users] Polycom IP 500 - MWI Not Working > > >Is tehre anyone out there with Polycom phones who has Message Waiting >Indicators working with the IP 500? >If so, can you tell me how you got it >working, what variable to set in * or the Polycom cfg files? > >Thanks! >W Just a thought, MWI doesn't work at this point with MYQLFRIENDS or res_data. I have a patch just about ready for res_data, but I am not quite there yet. Also, if you aren't populating from a database do you have [EMAIL PROTECTED] in your voicemail.conf? Hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 500 - MWI Not Working
Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this enough to help me find why I do not get the notification on the phone when a message is waiting. Is tehre anyone out there with Polycom phones who has Message Waiting Indicators working with the IP 500? If so, can you tell me how you got it working, what variable to set in * or the Polycom cfg files? Thanks! W
[Asterisk-Users] AstMan
Hello All, Does anyone know the state of AstMan? I found some information and source code in the archive but it is from November of 2003. There is mention of a lgpl release but nothing else after. I would like to code in some of the features that were lacking like setting this in system tray and using a popup message system but I do not want to step on anyone's toes. Any info woud be appreciated regarding this project and the proper protocol to contribute code. Thanks, Wiley Siler
RE: [Asterisk-Users] Polycom IP Soundpoint 600 & early dial
THat is part of your phone configuration file, not Asterisk. Look in cfg files for your sip phone. If should be in phone.cfg More info is in the Polycom manual under Dialplan and digitmap W From: Mike Roberts [mailto:[EMAIL PROTECTED] Sent: Thursday, July 29, 2004 4:47 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Polycom IP Soundpoint 600 & early dial Is anyone successfully using this phone with *? I have one, and it is an excellent phone. However, I cannot figure out how to make the phone "early dial" -- that is, automatically dial the number without the user having to press the send button. Any ideas? Thanks, Mike Roberts
RE: [Asterisk-Users] Play CD!
MP3s have to use constant bitrate not variable bit rate. Look in the documentation for mpg123. -Original Message- From: Jozeph Brasil [mailto:[EMAIL PROTECTED] Sent: Saturday, July 24, 2004 5:30 AM To: [EMAIL PROTECTED] Subject: RES: [Asterisk-Users] Play CD! I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. -Mensagem original- De: Chris Foster [mailto:[EMAIL PROTECTED] Enviada em: sábado, 24 de julho de 2004 01:37 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Play CD! On Fri, 23 Jul 2004 23:29:19 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi all, > > Is it possible to play a CD has MusicOnHold? > > Thanks, > Jozeph > Why don't you just rip the CD to MP3? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reinstalled FRom CVS - Things are really different now...
Hello All, I rebuilt my machine adn there has been about 2 weeks time since my original CVS checkout. I have seen teh changes for features.conf so that does nto worry me. Hwoever, after moving over my saved conf files, things are not really running. Does anyone know aht happened to teh dial command from the CLI prompt? I cannot make a dial session to test to/from phones. Are teh latest release changes documented somewhere? Thanks! Wiley
[Asterisk-Users] Call Quality - Factors and Config Values
Hello All, I have a system up and running that will be used as a PBX lcaolly with SIP phones. Because I am dumping all my calls into my X100Ps and have a very small number of clients (15), I woudl like to set all my call quality variables to their highest levels. I ahve a 100 meg network with a switch that has Class of Service so I have no bandwidth limitations. Can anyone tell me what values in which files I shoudl change in order to set all my settings to their highest call quality settings? Some that I know I have set are... sip.conf using g711 zapata.conf bandwidth=high Thanks! Wiley
[Asterisk-Users] ZAP Channel doesn't hang up - X100P
When receiving an incoming call, I get sent to my IVR just fine. My Playback event plays back my test file and the itis suppose to Hangup. The Hangup app fires off and I see the console say it hungup the line. However, I cannot receive anymore calls after that. When I run 'zap show channel 1' I see my zap channel in a state of off hook. The only thing that fixes it is stop/restart zaptel. Does anyone know why this may not be dropping the line correctly? Thanks! Wiley
RE: [Asterisk-Users] Polycom IP 500 Voicemail
John,
I got my config fixed. Needed to rerun make install in asterisk since
zaptel was setup again. Now on to the IVR tomorrow and trying to get
this vmail button working right.
Thanks!
Wiley
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 7:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
OK, let's work on this.
> Actually, I am having trouble with my X100P setup too which will >
probably sow when you read through my configs. I cannot get my >
referencing from contaxt to context setup correctly.
First things first. I would like to see how your phones are setup in
sip.conf along with your voicemail.conf. Specifically, what context the
sip phones are put into and whether or not the extensions of the sip
phones match your voicemail boxes.
For example, from my sip.conf file for my extension 7001, I have:
[7001]
context=from-internal
callerid="John Baker" <7001>
type=friend
username=7001
secret=X
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms
away
protocol=udp
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=0
disallow=all
allow=ulaw
allow=gsm
auth=md5
and the relevant line from voicemail.conf is
[listbrokers]
7001 => X,John Baker,[EMAIL PROTECTED],,tz=central
> Now I need to do something in oss.conf and zapata.conf to ensure
which > one answers the X100P right?
Yeah, this is a mess. First, are we answering phone calls on the
console? If yes, you're going to need your incoming phones to ring
/dev/console. I don't think you want this, so oss.conf can wait.
Second, why does your incoming context also include local and outgoing?
That doesn't seem to quite right to me.
And what is this?
> [outgoing]
> exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
John
Wiley E. Siler wrote:
> Actually, I am having trouble with my X100P setup too which will
> probably sow when you read through my configs. I cannot get my
> referencing from contaxt to context setup correctly.
>
>
> These are in extensions.conf
> ; --
> ; GLOBALS - Defines variables for use of devices, extensions ;
> --
>
> [globals]
> ;Reception
> PHONES0=SIP/2000
> PHONES0VM=2000
>
> PHONES1=SIP/2001
> PHONES1VM=2001
>
> PHONES2=SIP/2002
> PHONES2VM=2002
>
> PHONES3=SIP/2003
> PHONES3VM=2003
>
> ;Trunk Info
> TRUNK=Zap/g1 ; Trunk interface
> TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
>
> ; --
> ; END GLOBALS
> ; --
>
>
> [macro-vmessage]
> exten => s,1,VoiceMail2(u${ARG1})
> exten => s,2,Playback(groovy)
> ;exten => s,3,BackGround(dialing)
> exten => s,3,Playback(goodbye)
> exten => s,4,Hangup
>
> ; --
> ; DEFINE EXTENSIONS
> ; --
>
> [trunkint]
> ;
> ; International long distance through trunk
> ;
> exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9011.,2,Congestion
>
> [trunkld]
> ;
> ; Long distance context accessed through trunk
> ;
> exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91NXXNXX,2,Congestion
>
> [trunklocal]
> ;
> ; Local seven-digit dialing accessed through trunk interface
> ;
> exten => _9480XXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9480XXX,2,Congestion
>
> exten => _9602XXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9602XXX,2,Congestion
>
> exten => _9623XXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9623XXX,2,Congestion
>
>
> exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9NXX,2,Congestion
>
> [trunktollfree]
> ;
> ; Long distance context accessed through trunk interface
> ;
> exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91800NXX,2,Congestion
> exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXX,2,Congestion
> exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXX,2,Congestion
> exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXX,2,Congestion
>
> [international]
> ;
> ; Master context for international long distance
> ;
> ignorepat => 9
> include => longdi
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Voicemail.conf
-
[general]
format=gsm
[local]
2000 => 1234,Sarah,[EMAIL PROTECTED]
2001 => 1234,Gene,[EMAIL PROTECTED]
2002 => 1234,Lee,[EMAIL PROTECTED]
2003 => 1234,Wiley,[EMAIL PROTECTED]
--
Sip.conf
---
[general]
port=5060
[2000]
type=friend
host=dynamic
context=local
allow=g711
secret=PASSWORD
callerid="Front Desk" <2000>
mailbox=2000
dtmfmode=rfc2833
nat=0
[2001]
type=friend
context=local
allow=g711
secret=PASSWORD
callerid="Gene" <2001>
mailbox=2001
dtmfmode=rfc2833
nat=0
[2002]
type=friend
host=dynamic
context=local
allow=g711
secret=PASSWORD
callerid="Lee" <2002>
mailbox=2002
dtmfmode=rfc2833
nat=0
[2003]
type=friend
host=dynamic
context=local
;allow=g729
allow=g711
secret=PASSWORD
callerid="Wiley" <2003>
mailbox=2003
dtmfmode=rfc2833
nat=0
--
Other Stuff
--
And what is this?
> [outgoing]
> exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
Mistake on my part. I changed this to g1 which is correct, right?
> Now I need to do something in oss.conf and zapata.conf to
ensure which > one answers the X100P
right?
Yeah, this is a mess. First, are we answering phone calls on
the console? If yes, you're going to
need your incoming phones to ring /dev/console. I don't
think you want this, so oss.conf can wait.
I can honestly say. I have no idea. This is where the idea of contexts
breaks aoart for me. I want to start out just making my server pass the
ring to a group of phones (see setup in original mail). Later I am
going to define some IVR stuff and have * pick up the line and route to
people on user input.
--
Second, why does your incoming context also include local and
outgoing?
That doesn't seem to quite right to me.
I corrected it and I will continue to try and update.
Thanks for you help!
Wiley
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 7:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
OK, let's work on this.
> Actually, I am having trouble with my X100P setup too which will >
probably sow when you read through my configs. I cannot get my >
referencing from contaxt to context setup correctly.
First things first. I would like to see how your phones are setup in
sip.conf along with your voicemail.conf. Specifically, what context the
sip phones are put into and whether or not the extensions of the sip
phones match your voicemail boxes.
For example, from my sip.conf file for my extension 7001, I have:
[7001]
context=from-internal
callerid="John Baker" <7001>
type=friend
username=7001
secret=X
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms
away
protocol=udp
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=0
disallow=all
allow=ulaw
allow=gsm
auth=md5
and the relevant line from voicemail.conf is
[listbrokers]
7001 => X,John Baker,[EMAIL PROTECTED],,tz=central
> Now I need to do something in oss.conf and zapata.conf to ensure
which > one answers the X100P right?
Yeah, this is a mess. First, are we answering phone calls on the
console? If yes, you're going to need your incoming phones to ring
/dev/console. I don't think you want this, so oss.conf can wait.
Second, why does your incoming context also include local and outgoing?
That doesn't seem to quite right to me.
And what is this?
> [outgoing]
> exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
John
Wiley E. Siler wrote:
> Actually, I am having trouble with my X100P setup too which will
> probably sow when you read through my configs. I cannot get my
> referencing from contaxt to context setup correctly.
>
>
> These are in extensions.conf
> ; --
> ; GLOBALS - Defines variables for use of devices, extensions ;
> --
>
> [globals]
> ;Reception
> PHONES0=SIP/2000
> PHONES0VM=2000
>
> PHONES1=SIP/2001
> PHONES1VM=2001
>
> PHONES2=SIP/2002
> PHONES2VM=2002
>
> PHONES3=SIP/2003
> PHONES3VM=2003
>
> ;Trunk Info
> TRUNK=Zap/g1 ; Trunk interface
> TRUNKMSD=1 ; MSD digit
RE: [Asterisk-Users] Polycom IP 500 Voicemail
;
exten => 6001,1,Dial(${PHONES0}&${PHONES1}&${PHONES2}&${PHONES3},20,trf)
exten => 6001,2,Dial(SIP/6000,20,trf)
exten => 6001,3,Hangup
;---
; END RING EVERYONE
;--
;--
; DEFINE CALL PARKING AREA
;-
include =>parkedcalls
;--
; DEFINE MEETING ROOMS
;-
;exten => 4000,1,Meetme,4
exten => s,1,Answer
exten => s,2,BackGround(greeting)
exten => t,1,Playback(vm-goodbye)
exten => t,2,HangUp
[incoming]
exten => s,1,Answer
exten => s,2,Dial(SIP/2000)
exten => s,3,Hangup
include => local
include => outgoing
[outgoing]
exten => _9.,1,Dial(ZAP/g2/${EXTEN,1})
Now I need to do something in oss.conf and zapata.conf to ensure which
one answers the X100P right?
in zapata.conf...
[channels]
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.5
txgain=0.5
group=1
pickupgroup=1
immediate=no
signalling=fxs_ks
callerid=asreceived
channel=1
context=incoming
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 10:52 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
Mr. Siler -
I respond in kind...
> I am using the latest firmware from the Wiki. 2.4.2 I believe.
Oops. The latest firmware version is 1.2.0
Try http://www.freedomphones.net/polycom/files/ for the latest firmware.
If you don't show the latest version, (try pressing the right buttons
on your Polycom phone to get a version number) then anything else we
discuss is worthless.
> I edit my XML docs in notepad only.
DON'T DO THAT!! Trust me. I wasted alot of time with a text editor.
For a free XML editor, I use http://www.xmlcooktop.com/ Oh, and by the
way...
USING AN XML EDITOR IS VERY IMPORTANT!!! Polycom phones will load
corrupt XML, but not the way you want it. You will think your changes
have an effect, but if the XML isn't good, then they won't. Test your
settings with an XML editor!!! Make sure your config files read OK.
> This retrieves my mail through the menu system but not directly.
Directly to me means I press the 'Messages' button on my Polycom 600 and
asterisk asks me for a password. (Asterisk discerns the mailbox from
the extension of the phone) It's one touch (but still password
protected) It's working here and I'm sure we can get it to work at your
office.
> Voicemail answers on extension 8.
Just to be sure, can I see your extensions.conf?
John
Wiley E. Siler wrote:
> I have tried both a nul and the following...
>
> Subscribe = 8
> callbackmode = contact
> Callback = 8
>
> This retrieves my mail through the menu system but not directly.
>
> I am using the latest firmware from the Wiki. 2.4.2 I believe.
>
> I edit my XML docs in notepad only.
>
> Voicemail answers on extension 8.
>
> Thanks,
> Wiley
>
>
>
> -Original Message-
> From: John Baker [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 20, 2004 3:48 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
>
> Why do you have a non-null msg.mwi.1.subscribe? You're sending a
> SUBSCRIBE request to asterisk at extension '8' upon bootup. Is that
> what you want?
>
> Did you upgrade the phone with the latest firmware?
>
> Did you use an XML editor to mess with the configuration? I messed up
> mine once using a text editor.
>
> Is asterisk setup to answer voicemail at extension '8'?
>
> Try the above and let me know.
>
> John
>
> On Tue, 2004-07-20 at 11:52, Wiley E. Siler wrote:
>
>>I tried this configuration and it still does not work for me. In
>>fact, now I cannot dial in using the menu system of the message
>>center. Here is how I have now mine configured and what I get...
>>
>>
>> >msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="8"
>>msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="registration"
>>msg.mwi.2.callBack="" msg.mwi.3.subscribe=""
>>msg.mwi.3.callBackMode="registration" msg.mwi.3.callBack=""
>>msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="registration"
>>msg.mwi.4.callBack="" msg.mwi.5.subscribe=""
>>msg.mwi.5.callBackMode="registration" msg.mwi.5.callBack=""
>>msg.mwi.6.subscribe="" msg.mwi.6.call
RE: [Asterisk-Users] Installing X100P
That did it. I have the wcfxo running and channeled. Now I just have to beat my dial pan. I can dial internally to all my SIPs but outbound and inbound off the X100P are still not running. Do I just do this... Define [incoming] in extensions [incoming] exten => 1234567,1,Dial(SIP/2000) ; 1234567 = a local incoming call number? exten => 1234567,2,Congestion Is this correct? Thanks for the help! Wiley -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 7:41 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Installing X100P Install the kernel-source RPM off of the RH9 CD. -Seth On Tue, 2004-07-20 at 20:28, Wiley E. Siler wrote: > The error I receive when I run make > > Thanks, > Wiley > > > -Original Message- > From: Wiley E. Siler > Sent: Tuesday, July 20, 2004 4:12 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Installing X100P > > Could this have to do with the fact that I do not have a copy of the > redhat source code in the palce specified immediately at the top of > Makefile? The writer makes reference to Redhat breaking stuff and > that the headers... Here is is... > > # Okay, the people at RedHat have to break everything they can > possibly even attempt to. > # So, we have to look in /usr/src/linux-2.4/include for header files > given their brain dead # crappy installation. (Mind you, I'm a RedHat > user myself, so I suppose I'm just as # stupid as they are). Everyone > else who is mildly sane of course links /usr/include/linux # to their > working kernel source directory, the way God himself does, of course # > (assuming He's running Linux -- which we all know He must). > > > Well, I do not have a copy of those src files lcoated there. I > installed from Redha 9.0 cds. Do I need to get a copy of the linux > kernal source before I compile the zaptel stuff? > > Thanks, > Wiley > > > -Original Message- > From: Seth Remington [mailto:[EMAIL PROTECTED] > Sent: Tuesday, July 20, 2004 2:09 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Installing X100P > > You have to compile and install zaptel *before* asterisk for that to > work. You don't have to change your version, just "make install" in > zaptel source directory and then "make clean" & "make install" in > asterisk source directory. > > -Seth > > On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote: > > I attempted to install an X100P card but it was not correctly > > recognized by my Redhat 9 install. I had a test install running > > without any cards which was working great minus the outward dialing > > since no cards existed. Now that I have a card, I want to add it to > > the system. Do I have to scratch the whole current install in order > > to get the X100P running on my system or is there a way to get it > > installed as is? I really do not want to change my version of > > Asterisk since it is running well at this point. Is it possible to > > just update and add the card? > > > > Thanks, > > Wiley > > > -- > Seth Remington > SaberLogic, LLC > 661-B Weber Drive > Wadsworth, Ohio 44281 > Phone: (330)335-6442 > Fax: (330)336-8559 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
I have tried both a nul and the following... Subscribe = 8 callbackmode = contact Callback = 8 This retrieves my mail through the menu system but not directly. I am using the latest firmware from the Wiki. 2.4.2 I believe. I edit my XML docs in notepad only. Voicemail answers on extension 8. Thanks, Wiley -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 3:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Why do you have a non-null msg.mwi.1.subscribe? You're sending a SUBSCRIBE request to asterisk at extension '8' upon bootup. Is that what you want? Did you upgrade the phone with the latest firmware? Did you use an XML editor to mess with the configuration? I messed up mine once using a text editor. Is asterisk setup to answer voicemail at extension '8'? Try the above and let me know. John On Tue, 2004-07-20 at 11:52, Wiley E. Siler wrote: > I tried this configuration and it still does not work for me. In > fact, now I cannot dial in using the menu system of the message > center. Here is how I have now mine configured and what I get... > > >msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="8" > msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="registration" > msg.mwi.2.callBack="" msg.mwi.3.subscribe="" > msg.mwi.3.callBackMode="registration" msg.mwi.3.callBack="" > msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="registration" > msg.mwi.4.callBack="" msg.mwi.5.subscribe="" > msg.mwi.5.callBackMode="registration" msg.mwi.5.callBack="" > msg.mwi.6.subscribe="" msg.mwi.6.callBackMode="registration" > msg.mwi.6.callBack=""/> > > >up.oneTouchVoiceMail="1"/> > > > > The relevent fields being the msg. fields and up.oneTouchVoicemail > > This allows me voicemail via the Messages button but it is not direct. > I have to navigate still through allt he menus. > > W > > > > -Original Message- > From: John Baker [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004 10:17 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail > > My Polycom Message button goes straight to voicemail. Here's how: > > 1) Use the latest firmware, available on the Wiki > > 2) In your phone.cfg file (for each phone) set > > > msg.mwi.1.callBack="76" > > > 3) In your extensions.conf, have something like: > > exten => 76,1,VoiceMailMain2([EMAIL PROTECTED]) > > (Let's assume your voice mailbox is the same as your extension) > > Then when you push the message button, asterisk will ask for your > password! You're in! > > John > > > Chris A. Icide wrote: > > On 04:28 PM 7/19/2004, Wiley E. Siler wrote: > > >Mine does the same. Once in Message center I can choose selection > > >1.Message Center and then soft key Select.Then I select the > > >registered line that I want to check voice mail on. That is no > > less than > > >4 key strokes just to get into your voice mail. Not many to me > > but tons >to an unskilled user. However, in the documentation > > regarding the >bypassInstantMessage value, supposedly, setting > > bypassInstantMessage to > > >1 is supposed to allow you to go right into voice mail without > > >navigating the Message Center. That is the big question on my mind > > at >this point. I have yet to get this to work and I also don't > > think I am >receiving any SIMPLE messages ti show me that I have > messages waiting. > > > > > >Do you get a message waiting indicator? > > > > > > > I do get MWI, there are a few things you need to set, and I forget > > what off the top of my head, soon as I can look and post it here. > > > > I haven't tried the bypassInstantMessage value, but I'll take a look > > and see if I can get it to work. > > > > -Chris > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options vi
RE: [Asterisk-Users] Installing X100P
Could this have to do with the fact that I do not have a copy of the redhat source code in the palce specified immediately at the top of Makefile? The writer makes reference to Redhat breaking stuff and that the headers... Here is is... # Okay, the people at RedHat have to break everything they can possibly even attempt to. # So, we have to look in /usr/src/linux-2.4/include for header files given their brain dead # crappy installation. (Mind you, I'm a RedHat user myself, so I suppose I'm just as # stupid as they are). Everyone else who is mildly sane of course links /usr/include/linux # to their working kernel source directory, the way God himself does, of course # (assuming He's running Linux -- which we all know He must). Well, I do not have a copy of those src files lcoated there. I installed from Redha 9.0 cds. Do I need to get a copy of the linux kernal source before I compile the zaptel stuff? Thanks, Wiley -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 2:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Installing X100P You have to compile and install zaptel *before* asterisk for that to work. You don't have to change your version, just "make install" in zaptel source directory and then "make clean" & "make install" in asterisk source directory. -Seth On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote: > I attempted to install an X100P card but it was not correctly > recognized by my Redhat 9 install. I had a test install running > without any cards which was working great minus the outward dialing > since no cards existed. Now that I have a card, I want to add it to > the system. Do I have to scratch the whole current install in order > to get the X100P running on my system or is there a way to get it > installed as is? I really do not want to change my version of > Asterisk since it is running well at this point. Is it possible to > just update and add the card? > > Thanks, > Wiley > -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing X100P
I have tried this repeatedly and I get errors and no output. I tried with the CVS version and the download rfom ftp.digium.com. I have the output of the make command but it is 109k in text file. Can I post an email with a zip file or is that not allowed? Wiley s -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 2:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Installing X100P You have to compile and install zaptel *before* asterisk for that to work. You don't have to change your version, just "make install" in zaptel source directory and then "make clean" & "make install" in asterisk source directory. -Seth On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote: > I attempted to install an X100P card but it was not correctly > recognized by my Redhat 9 install. I had a test install running > without any cards which was working great minus the outward dialing > since no cards existed. Now that I have a card, I want to add it to > the system. Do I have to scratch the whole current install in order > to get the X100P running on my system or is there a way to get it > installed as is? I really do not want to change my version of > Asterisk since it is running well at this point. Is it possible to > just update and add the card? > > Thanks, > Wiley > -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing X100P
When I run make I get all kinds of errors. So far I ahve yet to get past that problem and when I look for /etc/zaptel.conf and /etc/asterisk/zaptel.com these fiels do not exist. W From: Celedonio Albarran [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 12:57 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Installing X100P Compile zaptel Edit /etc/zaptel.conf and /etc/asterisk/zaptel.conf modprobe zaptel modprobe wcfxo ztcfg start asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. SilerSent: Tuesday, July 20, 2004 1:55 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Installing X100P I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it installed as is? I really do not want to change my version of Asterisk since it is running well at this point. Is it possible to just update and add the card? Thanks, Wiley
[Asterisk-Users] Error on Zaptel install
I attempt to run make clean:make install and I get the following (cut short for brevity). zaptel.c: In function `zt_init':zaptel.c:6123: warning: implicit declaration of function `register_chrdev'zaptel.c:6124: `KERN_ERR' undeclared (first use in this function)zaptel.c:6124: parse error before string constantzaptel.c:6129: `KERN_INFO' undeclared (first use in this function)zaptel.c:6129: parse error before string constantzaptel.c:6134: warning: implicit declaration of function `rwlock_init'zaptel.c: In function `zt_cleanup':zaptel.c:6148: `KERN_INFO' undeclared (first use in this function)zaptel.c:6148: parse error before string constantzaptel.c:6167: warning: implicit declaration of function `unregister_chrdev'zaptel.c: At top level:zaptel.c:6021: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 I cannot modprobe wcfxo my card or even get this install to complete. I ahve been testin Asterisks without a fxo card so now that I want to add one, do I have to rebuild from scratch? Thanks, Wiley
[Asterisk-Users] Installing X100P
I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it installed as is? I really do not want to change my version of Asterisk since it is running well at this point. Is it possible to just update and add the card? Thanks, Wiley
RE: [Asterisk-Users] Polycom IP 500 Voicemail
I tried this configuration and it still does not work for me. In fact, now I cannot dial in using the menu system of the message center. Here is how I have now mine configured and what I get... The relevent fields being the msg. fields and up.oneTouchVoicemail This allows me voicemail via the Messages button but it is not direct. I have to navigate still through allt he menus. W -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail My Polycom Message button goes straight to voicemail. Here's how: 1) Use the latest firmware, available on the Wiki 2) In your phone.cfg file (for each phone) set 3) In your extensions.conf, have something like: exten => 76,1,VoiceMailMain2([EMAIL PROTECTED]) (Let's assume your voice mailbox is the same as your extension) Then when you push the message button, asterisk will ask for your password! You're in! John Chris A. Icide wrote: > On 04:28 PM 7/19/2004, Wiley E. Siler wrote: > >Mine does the same. Once in Message center I can choose selection > >1.Message Center and then soft key Select.Then I select the > >registered line that I want to check voice mail on. That is no less > than > >4 key strokes just to get into your voice mail. Not many to me but > tons >to an unskilled user. However, in the documentation regarding > the >bypassInstantMessage value, supposedly, setting > bypassInstantMessage to > >1 is supposed to allow you to go right into voice mail without > >navigating the Message Center. That is the big question on my mind > at >this point. I have yet to get this to work and I also don't > think I am >receiving any SIMPLE messages ti show me that I have messages waiting. > > > >Do you get a message waiting indicator? > > > > I do get MWI, there are a few things you need to set, and I forget > what off the top of my head, soon as I can look and post it here. > > I haven't tried the bypassInstantMessage value, but I'll take a look > and see if I can get it to work. > > -Chris > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo on a PRI
I think I saw a reference to a similar problem and it regarded IRQ issues on the machine in question. IF there was IRQ sharing, cagey things happened. But if the T1 card had a static IRQ, it resolved the issue. Does your T1 card have a dedicated IRQ? I am sure someone will be able to explain further and possibly give you some validation on your Mobo too? Thanks, Wiley -Original Message- From: David Goldfein [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo on a PRI Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Thank you so much! That was exactly what I needed to know! Cheersm Wiley -Original Message- From: Tor Roberts [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley, I don't have any 500s, but I use 600s, which use the same file I think. Here is my digitmap: What this says is that if I dial 9, then a 7 digit local number, I don't need to hit send. If I dial 91, then 10 digit long distance number, I don't need to hit send. If I dial extension 85 plus any 2 digits ex., 8523, I don't need to hit send. If I dial extension 5,6, or 7 plus 2 digits, ex. 635, I don't need to hit send. And if I dial 9411, or 9911 (info or emergency) I don't need to hit send. Hope this helps. -Tor Wiley E. Siler wrote: >I read the administrator document repeatedly. I have not been able to >find a wiki that applied to digitmap feature at all and I have searched >repeatedly and read several of the wikis regarding Polycoms. The >administrators guide doesn't have enough context explanation to make the >use of the digitmap understandable. > >That is the basis of my request for a digitmap explanation. I am not >asking someone to write mine for me. I am asking to see an example and >an explanation that gives context so I can write my own and know I have >done it properly. My PBX is Asterisk and the setup is about as generic >as generic can be. Polycoms over SIP to the PBX. > >If you know where the wiki is for digitmaps please send it. If you feel >inspired, a short explanation of the relevance and context of digitmaps >would be greatly appreciated. I know everyone has to take their own >time to answer these emails and I truly appreciate that. That is why I >do my research until I hit a wall, then I will ask here. I appreciate >whatever you can spare time for. > >Thanks! >Wiley > > > >-Original Message- >From: Brent Franks [mailto:[EMAIL PROTECTED] >Sent: Monday, July 19, 2004 10:26 AM >To: [EMAIL PROTECTED] >Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > > > >>Thank you! >> >>Can you tell me more about the dial plan feature? How do you setup >> >> >the > > >>correct digitmap? >> >> >> > >Check the Administrator's Document. You can find it on the Wiki, under >IP Phones.. Polycom. Did you try to look up the digitmap feature before >sending this post? If not, you should be able to understand it when you >read it, it's relatively straight forward. > >No one can setup a correct digitmap for you, as it will vary greatly on >how you have setup your PBX. > >- Brent > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages
Is this bascially setting your bandwith value = high inside of iax.conf? Or is there another place to designate the codec? Thanks, Wiley -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 2:11 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30 calls. I do have issues with processing CPU capacity. Is g711 CPU intensive as g729 ? I understand g729 is very CPU intensive. >>>... Forgive me, but what you just wrote tells you EXACTLY what you should use! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Mine does the same. Once in Message center I can choose selection 1.Message Center and then soft key Select.Then I select the registered line that I want to check voice mail on. That is no less than 4 key strokes just to get into your voice mail. Not many to me but tons to an unskilled user. However, in the documentation regarding the bypassInstantMessage value, supposedly, setting bypassInstantMessage to 1 is supposed to allow you to go right into voice mail without navigating the Message Center. That is the big question on my mind at this point. I have yet to get this to work and I also don't think I am receiving any SIMPLE messages ti show me that I have messages waiting. Do you get a message waiting indicator? W -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On 12:40 PM 7/19/2004, Wiley E. Siler wrote: >My Polycom is on loan as a demo and I assume it is one of the first >revision models. In fact it shows as Rev A on the back of the phone. > >I have all the same buttons you listed save for the Messages button. >The 3rd from the bottom on the right column of buttons sayd Voice Mail >on my version. That corresponds to the location of your button that >says Messages. I assume this was changed by Polycom since their phone >has other messaging capability (isntant message for instance) and it was >easier to use Messages and unify the meaning instead of Voice Mail and >lock it into one type of messaging. > >Does your Messages button dump you right into voice mail or do you have >to navigate a menu first? > >Thanks, >Wiley My messages button dumps me right to message center, which I then have to use soft buttons. My IP500 is Rev. C -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
And it is throughly convoluted in the admin guide. What is the T for? Pipe obviously separates entries. X = any digit one would assume? I am just luooking for a brief explanation. Thanks. Here is the excerpt from the manual. Attribute dialplan.digitmap Permitted Values string compatible with the digit map feature of MGCP described in 2.1.5 of RFC 3435. String is limited to 512 bytes and 20 segments; a comma is also allowed; when reached in the digit map, a comma will turn dial tone back on. [2-9]11|0T| 011xxx.T| [0-1][2- 9]x| [2-9]x| [2-9]xxxT Default Interpretation When this attribute is present, number-only dialing during the setup phase of new calls will be compared against the patterns therein and if a match is found, the call will be initiated automatically eliminating the need to press Send. Attribute Permitted Values Default Interpretation -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote: > Thank you! > > Can you tell me more about the dial plan feature? How do you setup the > correct digitmap? It is all in the Admin Guide you can download from the Polycom web site. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
My Polycom is on loan as a demo and I assume it is one of the first revision models. In fact it shows as Rev A on the back of the phone. I have all the same buttons you listed save for the Messages button. The 3rd from the bottom on the right column of buttons sayd Voice Mail on my version. That corresponds to the location of your button that says Messages. I assume this was changed by Polycom since their phone has other messaging capability (isntant message for instance) and it was easier to use Messages and unify the meaning instead of Voice Mail and lock it into one type of messaging. Does your Messages button dump you right into voice mail or do you have to navigate a menu first? Thanks, Wiley -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:46 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Strange, I have an IP500 right out of the new-plastic-gadget-smell box, and it doesn't have a button labelled Voicemail. On the left side are the blue speaker, red mute, and blue headset buttons, then next to them top to bottom are the three Line buttons (clear covers for putting your own labels), Directories, Services, Call Lists, Conference, Transfer, and Redial. On the right of the system, top side are the 4 way selection pad with select and delete, then below that are Menu, Messages, and Do Not Disturb, and finally Hold. In the middle are the 12 keypad keys, 4 soft keys, and volume + and - buttons. No where am I able to find a hard voicemail button. -Chris On 10:42 AM 7/19/2004, Wiley E. Siler wrote: >Thank you! > >Can you tell me more about the dial plan feature? How do you setup the >correct digitmap? > >W > >-Original Message- >From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] >Sent: Monday, July 19, 2004 4:56 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail > >Wiley E. Siler wrote: > >> I have a solution that allows me to assign a soft key with no >problems. >> However, it seems like a waste the the hard button labeled Voice Mail >> is not dialing right into voice mail. Is there a known way yo do >> this? I have tried everything in the manual but it doesn't seem to >> work. I have IP 500s and I want to be able to use all three display >> lines for just lines on the phone. >> >I think that feature is inly available on the 1.2.0 sip firmware. It >works on ours but when you press it, you still have to pick a line, then >connect. The line button goes right to the voicemail. > >> Also, do you know if it is possible to program the buttons along the >> bottom of the screen like normal soft buttons? >> >Probably, but I haven't looked into it enough > >> And finally... >> Is there a way to make the system dial without having to hit the Send >> key after dialing a number? >> >look at the digitmap in sip.cfg > >-rb > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
I read the administrator document repeatedly. I have not been able to find a wiki that applied to digitmap feature at all and I have searched repeatedly and read several of the wikis regarding Polycoms. The administrators guide doesn't have enough context explanation to make the use of the digitmap understandable. That is the basis of my request for a digitmap explanation. I am not asking someone to write mine for me. I am asking to see an example and an explanation that gives context so I can write my own and know I have done it properly. My PBX is Asterisk and the setup is about as generic as generic can be. Polycoms over SIP to the PBX. If you know where the wiki is for digitmaps please send it. If you feel inspired, a short explanation of the relevance and context of digitmaps would be greatly appreciated. I know everyone has to take their own time to answer these emails and I truly appreciate that. That is why I do my research until I hit a wall, then I will ask here. I appreciate whatever you can spare time for. Thanks! Wiley -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > Thank you! > > Can you tell me more about the dial plan feature? How do you setup the > correct digitmap? > Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? W -Original Message- From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 4:56 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote: > I have a solution that allows me to assign a soft key with no problems. > However, it seems like a waste the the hard button labeled Voice Mail > is not dialing right into voice mail. Is there a known way yo do > this? I have tried everything in the manual but it doesn't seem to > work. I have IP 500s and I want to be able to use all three display > lines for just lines on the phone. > I think that feature is inly available on the 1.2.0 sip firmware. It works on ours but when you press it, you still have to pick a line, then connect. The line button goes right to the voicemail. > Also, do you know if it is possible to program the buttons along the > bottom of the screen like normal soft buttons? > Probably, but I haven't looked into it enough > And finally... > Is there a way to make the system dial without having to hit the Send > key after dialing a number? > look at the digitmap in sip.cfg -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Control Script
Does anyone know where I can find a list of all the control scripts? I want to write a standard windows tool that will allow you to pregenerate the configuration for your Asterisk install and them press one button to have it log into your box and upload the scripts. Of course, I will let everyone know when it is complete. Thanks, Wiley
RE: [Asterisk-Users] Polycom IP 500 Voicemail
I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? Thanks for the tips! Wiley -Original Message- From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 5:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote: > Hello All, > I have some Polycom IP 500 phones that I would like to have configured > for direct dialing to our voice mail system. So far I have been > unable to get the hard button labeled Voice Mail to connect to > Asterisk without first passing through the message center prompts. I > have followed all the Admin Guide instructions regarding the phones > .cfg files and using up.bypassInstantMessage="1" > up.oneTouchVoicemail="1" in the XML to no avail. Has anyone been able > to get a Polycom 500 to use the hardbutton to retrieve voice mail and > drop directly into voice mail without going through all the menus? > We programmed line 3 (line 6 on the IP 600s) on each phone with its own context/registration and set the IP 500 to auto dial into voicemail. extensions.conf: [voicemail] exten => 5501,1,voicemailmain2,[EMAIL PROTECTED] The phone.cfg file has a setting for autodial. I assume you can get a phone registered, but make sure dtmfmode is set to inband and set a mailbox= line to get MWI working. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 500 Voicemail
Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using up.bypassInstantMessage="1" up. in the XML to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail and drop directly into voice mail without going through all the menus? Thanks, Wiley
RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
I just started out too and I can tell you it is easier to start from scratch with a good wiki then alter the demo files. Here is a wiki you can build a good working system with... http://www.wlug.org.nz/AsteriskSampleSetup For your ciscos search http://asterisk.xvoip.com/index.php Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 5:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a "extension not found in local" message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a "not found 404" messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: "11"; Line 1 Extension\User ID line1_displayname: "Lounge1"; Line 1 Display Name line1_authname: "lounge11" ; Line 1 Registration Authentication line1_password: "lounge"; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "137.222.10.60" ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: GMT ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when BST is in effect dst_start_month: April ; Month in which BST starts dst_start_day: "21" ; Day of month in which BST starts dst_start_day_of_week: Sun ; Day of week in which BST starts dst_start_week_of_month: 1 ; Week of month in which BST starts dst_start_time: 02 ; Time of day in which BST starts dst_stop_month: Oct ; Month in which BST stops dst_stop_day: "20" ; Day of month in which BST stops dst_stop_day_of_week: Sunday ; Day of week in which BST stops dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week of month dst_stop_time: 2 ; Time of day in which BST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic adjustment time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 proxy_backup: "" ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ;
RE: [Asterisk-Users] Asterisk NAT spa-2000
I would comment out these lines in sip.conf ;externip=111.222.333.444 ;localnet=192.168.1.0 ;localmask=255.255.255.0 Then set nat=no -Original Message- From: Simon Chappell [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 4:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk NAT spa-2000 Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 [EMAIL PROTECTED] context=sip callerid="James" <2001> secret=hidden canreinvite=no allow=ulaw nat=yes qualify=yes I added the nat and qualify entries after hunting round google but still get this error, spot the no nat bit. to 81.178.77.67:34504 Retransmitting #2 (no NAT): OPTIONS sip:81.178.77.67:34504 SIP/2.0 Via: SIP/2.0/UDP 62.188.201.123:5060;branch=z9hG4bK68af34fa From: "asterisk" ;tag=as5582cfae To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 18 Jul 2004 12:43:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 any ideas anyone thanks in advance Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 500 Phones - Button Assignment
Hello All, So far I have been unable to get the hard button labeled Voice Mail to conenct to Asterisk. I have followed all the Admin Guide instructions regarding the .cfg files and using up.bypassInstantMessage="1" up. to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail? ^Thanks, Wiley
RE: [Asterisk-Users] Using a group variable for a groupofextension to dial
Actually doing both sounds good to me. Can you explain further about
ringing them all at once?
Here is how I tried to make mine work and failed...
{global}
PHONES0=SIP/2000
PHONES1=SIP/2001
[local]
exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf)
When I dial 6001 I see my debugger tell me that I am using the wrong
syntax.
Do you know the correct syntax for ringing them all at once?
I will check out the queue system you referenced.
Thanks!
Wiley
-Original Message-
From: Seth Remington [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 17, 2004 7:45 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Using a group variable for a
groupofextension to dial
It would ring all three at the same time. You are probably looking for a
roundrobin call queue ->
http://www.voip-info.org/wiki-Asterisk+call+queues
I have never set up a call queue myself so I can't help you more than
pointing you to the link. You could also do what you want with just the
dialplan...
exten => 501,1,Dial(SIP/Phone1,10)
exten => 501,2,Dial(SIP/Phone2,10)
exten => 501,3,Dial(SIP/Phone3,10)
That would ring phone 1 for ten seconds, if it wasn't answered it would
ring phone 2 for ten seconds, etc...
-Seth
On Sat, 2004-07-17 at 22:05, Wiley E. Siler wrote:
> That could be it. What I want to do is set a group of callers and
> have the event cause the phone to ring them in order. I will tie it
> to my IVR portion and thus I can make sure peole in sales get calls
> based on our hierarchy in the office. So if I am reading your example
> right the syntax is
>
> Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf)
>
> Is that a valid way to cause it to ring through each of these
> extensions or would that result in these three extensions all ringing
togeher?
>
> Thanks!
> Wiley
>
>
>
> -Original Message-
> From: Seth Remington [mailto:[EMAIL PROTECTED]
> Sent: Saturday, July 17, 2004 6:35 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Using a group variable for a group
> ofextension to dial
>
> Maybe I am misunderstanding your question but are you looking for the
> '&' operator?
>
> Dial(type1/identifier1&type2/identifier2&type3/identifier3...,timeout,
> op
> tions,URL)
>
> -Seth
>
> On Sat, 2004-07-17 at 19:24, Wiley E. Siler wrote:
> > I ahve been searching to no avail for a referenc eon how to setup a
> > part of my dial plan that will ring certain groups of number based
> > upon the context. Essentually, I want to be able to designate 3
> > people as sales and have my IVR handoff and ring their extensions in
> > order. Then maybe I will ahve a couple of people I group together
> > and
>
> > have them ring if someone selects 2 on the IVR for tech support.
> > Someone used some sytax that employed something like ($GROUP1,
> > GROUP2,
> > GROUP3) that accomplished this but I cannot find any reference to it
> > after googling since ysterday. Does anyone know how to accomplish
> > this task?
> >
> > Thanks,
> > Wiley
> >
> --
> Seth Remington
> SaberLogic, LLC
> 661-B Weber Drive
> Wadsworth, Ohio 44281
> Phone: (330)335-6442
> Fax: (330)336-8559
>
> ___
> Asterisk-Users mailing list
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--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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RE: [Asterisk-Users] Using a group variable for a group ofextension to dial
That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause it to ring through each of these extensions or would that result in these three extensions all ringing togeher? Thanks! Wiley -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Saturday, July 17, 2004 6:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Using a group variable for a group ofextension to dial Maybe I am misunderstanding your question but are you looking for the '&' operator? Dial(type1/identifier1&type2/identifier2&type3/identifier3...,timeout,op tions,URL) -Seth On Sat, 2004-07-17 at 19:24, Wiley E. Siler wrote: > I ahve been searching to no avail for a referenc eon how to setup a > part of my dial plan that will ring certain groups of number based > upon the context. Essentually, I want to be able to designate 3 > people as sales and have my IVR handoff and ring their extensions in > order. Then maybe I will ahve a couple of people I group together and > have them ring if someone selects 2 on the IVR for tech support. > Someone used some sytax that employed something like ($GROUP1, GROUP2, > GROUP3) that accomplished this but I cannot find any reference to it > after googling since ysterday. Does anyone know how to accomplish > this task? > > Thanks, > Wiley > -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using a group variable for a group of extension to dial
I ahve been searching to no avail for a referenc eon how to setup a part of my dial plan that will ring certain groups of number based upon the context. Essentually, I want to be able to designate 3 people as sales and have my IVR handoff and ring their extensions in order. Then maybe I will ahve a couple of people I group together and have them ring if someone selects 2 on the IVR for tech support. Someone used some sytax that employed something like ($GROUP1, GROUP2, GROUP3) that accomplished this but I cannot find any reference to it after googling since ysterday. Does anyone know how to accomplish this task? Thanks, Wiley
RE: [Asterisk-Users] [OT] The stories people tell to support.
I think that is also an ID ten T problem. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, July 15, 2004 8:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] [OT] The stories people tell to support. On 15 Jul 2004 at 20:48, Dave Cotton wrote: > This one is for the archives. > > I got a call today that the * at one of my clients was not working. > The switchboard is set up to ring for a while and then the rest of the > phones start up if the switchboard doesn't pick up. This was not > happening. Instead the mobile phone of one of the people there was > ringing and after the delay the internals started ringing. > > When I connected to the web interface of the SNOM 200 I found the > redirection set to always and the number of the mobile proceeded by > the 0 to dial out as the destination. Apparently this is a bug with > the SNOM 200. If you move it 30cm., they didn't say in which > direction, it automatically chooses a phone number and sets up the > redirection. > > It's not 1st April is it? > You obviously didn't read the EULA...it says that they can install software on your phone or redirect it to numbers they have sniffed off your network. Luckily, this only happens between 3:45pm and 3:46pm on the 32nd day of the month. And only every third month. This feature is called ALWANCAS (annoying lusers who are never clear about stuff) Matt Riddell > -- > Dave Cotton <[EMAIL PROTECTED]> > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Updated Grandstream configurator
Steve, For your batch, be sure to include the /s switch after the command so it runs silent (no prompts) Thanks, Wiley -Original Message- From: Stephen R. Besch [mailto:[EMAIL PROTECTED] Sent: Thursday, July 15, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Updated Grandstream configurator Steve wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote: > >>The most recent version of GSConfigure is available at >>www.buffalo.edu/~sbesch Several serious bugs that kept the program >>from getting started have been ferreted out and corrected with the >>help of Bruce Komito. The program is now actually running on someone's >>machine other than mine. I have built this version with the oldest >>copies of the system dll's that I could find inn an effort to solve >>the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. >>You should have at least SP3, or even better, SP4 on Win2k. I believe >>it will run on Win9x, but I have not tested it and can make no guarantees. >> >>Steve Besch > > > The bad part is that starting with SP2 on w2k ms EULA has changed to > include your agreement to let microsoft not only see, what you have on > your computer, but also install software on it. This has caused a big > corporate hold on updating beyond SP2. The medical industry in > particular is having a hard time, as ms has not signed a non > disclosure to have access to personal medical information. > > - -- > Steve > > "They that would give up essential liberty for temporary safety > deserve neither liberty nor safety." > Benjamin Franklin > Since I am quite sure that the program will run without updating any of the dll's, what I should do is simply register them with regsvr32 from a batch job and bag the VB6 installer altogether. Before I do that though, can anyone tell me if regsvr32 ships with standard Win2k/WinXP? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Updated Grandstream configurator
It absolutely ships with Windows 2K/XP versions. Regsvr32 will work from any folder on a standard install. Wiley -Original Message- From: Stephen R. Besch [mailto:[EMAIL PROTECTED] Sent: Thursday, July 15, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Updated Grandstream configurator Steve wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote: > >>The most recent version of GSConfigure is available at >>www.buffalo.edu/~sbesch Several serious bugs that kept the program >>from getting started have been ferreted out and corrected with the >>help of Bruce Komito. The program is now actually running on someone's >>machine other than mine. I have built this version with the oldest >>copies of the system dll's that I could find inn an effort to solve >>the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. >>You should have at least SP3, or even better, SP4 on Win2k. I believe >>it will run on Win9x, but I have not tested it and can make no guarantees. >> >>Steve Besch > > > The bad part is that starting with SP2 on w2k ms EULA has changed to > include your agreement to let microsoft not only see, what you have on > your computer, but also install software on it. This has caused a big > corporate hold on updating beyond SP2. The medical industry in > particular is having a hard time, as ms has not signed a non > disclosure to have access to personal medical information. > > - -- > Steve > > "They that would give up essential liberty for temporary safety > deserve neither liberty nor safety." > Benjamin Franklin > Since I am quite sure that the program will run without updating any of the dll's, what I should do is simply register them with regsvr32 from a batch job and bag the VB6 installer altogether. Before I do that though, can anyone tell me if regsvr32 ships with standard Win2k/WinXP? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 500 and Asterisk
Hello All, Thanks for all the great info! Is there anyone out there using Polycom IP 500 phones with Asterisk who can advise on how to get these phones easily configured? So far, I have ben unable to google up any tools for them and the example config files I have do not have any documentation. Any hep would be appreciated. Thanks all, Wiley
RE: [Asterisk-Users] Problem with multiple phones behind firewall
Do you have these values set? externip localnet localmask -Original Message- From: Harold Workman [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 1:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem with multiple phones behind firewall Hi, I am having a problem when I add multiple phones behind a Symmetric Firewall. Heres my situation. 11am - Phone A registers with * 11:01am - test call to Phone A. Call works fine. 11:02am - Phone B registers with * 11:03am - test call to Phone A fails, test call to phone B works fine. 11:04am - test call from Phone A to Phone B and vice versa works fine. 11:05am - Phone A re-registers with *. Test call to Phone A works fine now. This happens on almost all occasions. When I see one phone register behind a firewall, i then see the "Retransmitting #5 (NAT):" messages, until I received the "Jul 13 15:11:09 WARNING[1133718080]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)" I have nat=yes in my sip.conf file. I have tried using the qualify command, but I have never been able to get it to work behind a symmetric firewall to both a unknown sip phone and xlite. The moment I turn on qualify, I see the Options request sent out, and on the client see the options request, but I never see a response on * from the clients. Here is what my sip.conf looks like... [general] port = 5060 bindaddr = 64.72.107.10 context = exten maxexpirey=3000 defaultexpirey=300 disallow=all allow=alaw allow=ulaw [123456] type=friend secret=k3v1n nat=yes canreinvite=no host=dynamic dtmfmode=rfc2833 context=cytelmain [789012] type=friend secret=cytel nat=yes canreinvite=no host=dynamic dtmfmode=rfc2833 context=cytelmain What else is there for me to try to resolve my NAT problem with multiple users behind a symmetric firewall? Thanks, Harold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
