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? Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040718/2030d860/attachment.htm
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,s${CALLERID}@vmcontext The phone.cfg file has a setting for autodial. I assume you can get a phone registered, but make sure dtmfmode is set to inband and set a mailbox= line to get MWI working. -rb
I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? Thanks for the tips! Wiley -----Original Message----- From: Russ Beaupre, P.E. [mailto:russ@botech.net] Sent: Sunday, July 18, 2004 5:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote:> Hello All, > I have some Polycom IP 500 phones that I would like to have configured> for direct dialing to our voice mail system. So far I have been > unable to get the hard button labeled Voice Mail to connect to > Asterisk without first passing through the message center prompts. I > have followed all the Admin Guide instructions regarding the phones > .cfg files and using up.bypassInstantMessage="1" > up.oneTouchVoicemail="1" in the XML to no avail. Has anyone been able> to get a Polycom 500 to use the hardbutton to retrieve voice mail and > drop directly into voice mail without going through all the menus? >We programmed line 3 (line 6 on the IP 600s) on each phone with its own context/registration and set the IP 500 to auto dial into voicemail. extensions.conf: [voicemail] exten => 5501,1,voicemailmain2,s${CALLERID}@vmcontext 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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:russ@botech.net] Sent: Monday, July 19, 2004 4:56 AM To: asterisk-users@lists.digium.com 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 noproblems.> 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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."
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:mwless@mindworks.net] Sent: Monday, July 19, 2004 10:26 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail> Thank you! > > Can you tell me more about the dial plan feature? How do you setupthe> 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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:chris@netgeeks.net] Sent: Monday, July 19, 2004 11:46 AM To: asterisk-users@lists.digium.com 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:russ@botech.net] >Sent: Monday, July 19, 2004 4:56 AM >To: asterisk-users@lists.digium.com >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 >Asterisk-Users@lists.digium.com >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 >Asterisk-Users@lists.digium.com >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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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]xxxxxxxxx| [2-9]xxxxxxxxx| [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:eric@fnords.org] Sent: Monday, July 19, 2004 11:50 AM To: asterisk-users@lists.digium.com 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 setupthe> 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
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:chris@netgeeks.net] Sent: Monday, July 19, 2004 3:03 PM To: asterisk-users@lists.digium.com 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 itwas >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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Thank you so much! That was exactly what I needed to know! Cheersm Wiley -----Original Message----- From: Tor Roberts [mailto:voip@sscsinc.com] Sent: Monday, July 19, 2004 3:35 PM To: asterisk-users@lists.digium.com 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: <digitmap dialplan.digitmap="9[2-9]xxxxxx|91xxxxxxxxxx|85xx|[5-7]xx|9411|9911" dialplan.digitmap.timeOut="3"/> 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 makethe>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 youfeel>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:mwless@mindworks.net] >Sent: Monday, July 19, 2004 10:26 AM >To: asterisk-users@lists.digium.com >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 featurebefore>sending this post? If not, you should be able to understand it whenyou>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 >Asterisk-Users@lists.digium.com >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 >Asterisk-Users@lists.digium.com >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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 msg.bypassInstantMessage="1">
<mwi msg.mwi.1.subscribe="8"
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=""/>
</msg>
<nat nat.ip="" nat.signalPort=""
nat.mediaPortStart=""/>
<user_preferences up.headsetMode="0"
up.useDirectoryNames="0"
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:JohnB@listbrokers.com]
Sent: Monday, July 19, 2004 10:17 PM
To: asterisk-users@lists.digium.com
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 msg.bypassInstantMessage="1">
<mwi msg.mwi.1.subscribe=""
msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="76" .... >
3) In your extensions.conf, have something like:
exten => 76,1,VoiceMailMain2(${EXTEN}@whatever_you_have_here)
(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
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
_______________________________________________
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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:johnb@listbrokers.com] Sent: Tuesday, July 20, 2004 3:48 PM To: asterisk-users@lists.digium.com 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 msg.bypassInstantMessage="1"> > <mwi msg.mwi.1.subscribe="8" > 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=""/> > </msg> > <nat nat.ip="" nat.signalPort="" nat.mediaPortStart=""/> > <user_preferences up.headsetMode="0" up.useDirectoryNames="0" > 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:JohnB@listbrokers.com] > Sent: Monday, July 19, 2004 10:17 PM > To: asterisk-users@lists.digium.com > 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 msg.bypassInstantMessage="1"> > <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" > msg.mwi.1.callBack="76" .... > > > 3) In your extensions.conf, have something like: > > exten => 76,1,VoiceMailMain2(${EXTEN}@whatever_you_have_here) > > (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 > > Asterisk-Users@lists.digium.com > > 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 > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9480XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9480XXXXXXX,2,Congestion
exten => _9602XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9602XXXXXXX,2,Congestion
exten => _9623XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9623XXXXXXX,2,Congestion
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => trunklocal
include => trunktollfree
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password@bigserver/local
; now if we dial 8, we can check voicemail.
;
;exten => 8,1,VoicemailMain(s${CALLERIDNUM}) ;= no password
exten => 8,1,VoicemailMain(${CALLERIDNUM}) ;= password
exten => 8,2,Hangup
; Add some more extensions for the two extensions . now we'll be able to
call one extension from the other.
; And if no one answers, it will go to the mailbox for that extension.
;
; extension 2000
;
exten => 2000,1,Dial(${PHONES1},20,trf)
exten => 2000,2,Macro(vmessage,${PHONES0VM})
exten => 2000,3,Hangup
;
; extension 2001
;
exten => 2001,1,Dial(${PHONES1},20,trf)
exten => 2001,2,Macro(vmessage,${PHONES1VM})
exten => 2001,3,Hangup
;
; extension 2002
;
exten => 2002,1,Dial(${PHONES2},20,trf)
exten => 2002,2,Macro(vmessage,${PHONES2VM})
exten => 2002,3,Hangup
;
; extension 2003
;
exten => 2003,1,Dial(${PHONES3},20,trf)
exten => 2003,2,Macro(vmessage,${PHONES3VM})
exten => 2003,3,Hangup
;
; IVR CHOICE 1
;
exten => 1,1,Answer
exten => 1,2,Playback(tt-somethingwrong)
exten => 1,3,Playback(tt-monkeysintro)
exten => 1,4,Playback(tt-monkeys)
exten => 1,5,Hangup
;
; IVR RECORDER
;
; Record voice file to /tmp directory
exten => 205,1,Wait(2) ; Call 205 to Record new Sound Files
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,wait(2)
exten => 205,6,Hangup
;
;MUSIC ON HOLD EXTENSION
;
exten => 6000,1,Answer
exten => 6000,2,SetMusicOnHold(default)
exten => 6000,3,MusicOnHold()
exten => 6000,4,Hangup
; ---------------------------------------------
; END DEFINE EXTENSIONS
; ----------------------------------------------
;------------------------------------
; RING EVERYONE EXTENSIONS
;-----------------------------------
;
; ring everyone
;
exten =>
6001,1,Dial(${PHONES0}&${PHONES1}&${PHONES2}&${PHONES3},20,trf)
exten => 6001,2,Dial(SIP/6000,20,trf)
exten => 6001,3,Hangup
;-----------------------------------
; END RING EVERYONE
;----------------------------------
;----------------------------------------------
; DEFINE CALL PARKING AREA
;---------------------------------------------
include =>parkedcalls
;--------------------------------------------------
; DEFINE MEETING ROOMS
;-------------------------------------------------
;exten => 4000,1,Meetme,40000
exten => s,1,Answer
exten => s,2,BackGround(greeting)
exten => t,1,Playback(vm-goodbye)
exten => t,2,HangUp
[incoming]
exten => s,1,Answer
exten => s,2,Dial(SIP/2000)
exten => s,3,Hangup
include => local
include => outgoing
[outgoing]
exten => _9.,1,Dial(ZAP/g2/${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:JohnB@listbrokers.com]
Sent: Tuesday, July 20, 2004 10:52 PM
To: asterisk-users@lists.digium.com
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:johnb@listbrokers.com]
> Sent: Tuesday, July 20, 2004 3:48 PM
> To: asterisk-users@lists.digium.com
> 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 msg.bypassInstantMessage="1">
>> <mwi msg.mwi.1.subscribe="8"
>>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=""/>
>> </msg>
>> <nat nat.ip="" nat.signalPort=""
nat.mediaPortStart=""/>
>> <user_preferences up.headsetMode="0"
up.useDirectoryNames="0"
>>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:JohnB@listbrokers.com]
>>Sent: Monday, July 19, 2004 10:17 PM
>>To: asterisk-users@lists.digium.com
>>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 msg.bypassInstantMessage="1">
>><mwi msg.mwi.1.subscribe=""
msg.mwi.1.callBackMode="contact"
>>msg.mwi.1.callBack="76" .... >
>>
>>3) In your extensions.conf, have something like:
>>
>>exten => 76,1,VoiceMailMain2(${EXTEN}@whatever_you_have_here)
>>
>>(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
>>>Asterisk-Users@lists.digium.com
>>>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
>>Asterisk-Users@lists.digium.com
>>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
>>Asterisk-Users@lists.digium.com
>>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
> Asterisk-Users@lists.digium.com
> 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
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
_______________________________________________
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Voicemail.conf
-------------------------------------------------------------
[general]
format=gsm
[local]
2000 => 1234,Sarah,123@456.com
2001 => 1234,Gene,123@456.com
2002 => 1234,Lee,123@456.com
2003 => 1234,Wiley,123@456.com
----------------------------------------------------------------------
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:JohnB@listbrokers.com]
Sent: Wednesday, July 21, 2004 7:04 PM
To: asterisk-users@lists.digium.com
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=XXXXXXXXX
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms
away
protocol=udp
dtmfmode=rfc2833
mailbox=7001@dont_spam_me
nat=0
disallow=all
allow=ulaw
allow=gsm
auth=md5
and the relevant line from voicemail.conf is
[listbrokers]
7001 => XXXXX,John Baker,JohnB@dont_spam_me.com,,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:${TRUNKMSD}})
> exten => _9011.,2,Congestion
>
> [trunkld]
> ;
> ; Long distance context accessed through trunk
> ;
> exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91NXXNXXXXXX,2,Congestion
>
> [trunklocal]
> ;
> ; Local seven-digit dialing accessed through trunk interface
> ;
> exten => _9480XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9480XXXXXXX,2,Congestion
>
> exten => _9602XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9602XXXXXXX,2,Congestion
>
> exten => _9623XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9623XXXXXXX,2,Congestion
>
>
> exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9NXXXXXX,2,Congestion
>
> [trunktollfree]
> ;
> ; Long distance context accessed through trunk interface
> ;
> exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91800NXXXXXX,2,Congestion
> exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXXXXXX,2,Congestion
> exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXXXXXX,2,Congestion
> exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXXXXXX,2,Congestion
>
> [international]
> ;
> ; Master context for international long distance
> ;
> ignorepat => 9
> include => longdistance
> include => trunkint
>
> [longdistance]
> ;
> ; Master context for long distance
> ;
> ignorepat => 9
> include => local
> include => trunkld
>
> [local]
> ;
> ; Master context for local, toll-free, and iaxtel calls only
> ;
> ignorepat => 9
> include => trunklocal
> include => trunktollfree
> ;
> ; You can use an alternative switch type as well, to resolve
> ; extensions that are not known here, for example with remote
> ; IAX switching you transparently get access to the remote
> ; Asterisk PBX
> ;
> ; switch => IAX2/user:password@bigserver/local
>
> ; now if we dial 8, we can check voicemail.
> ;
>
> ;exten => 8,1,VoicemailMain(s${CALLERIDNUM}) ;= no password
> exten => 8,1,VoicemailMain(${CALLERIDNUM}) ;= password
> exten => 8,2,Hangup
>
> ; Add some more extensions for the two extensions . now we'll be able
to> call one extension from the other.
> ; And if no one answers, it will go to the mailbox for that extension.
> ;
> ; extension 2000
> ;
> exten => 2000,1,Dial(${PHONES1},20,trf)
> exten => 2000,2,Macro(vmessage,${PHONES0VM})
> exten => 2000,3,Hangup
> ;
> ; extension 2001
> ;
> exten => 2001,1,Dial(${PHONES1},20,trf)
> exten => 2001,2,Macro(vmessage,${PHONES1VM})
> exten => 2001,3,Hangup
> ;
> ; extension 2002
> ;
> exten => 2002,1,Dial(${PHONES2},20,trf)
> exten => 2002,2,Macro(vmessage,${PHONES2VM})
> exten => 2002,3,Hangup
> ;
> ; extension 2003
> ;
> exten => 2003,1,Dial(${PHONES3},20,trf)
> exten => 2003,2,Macro(vmessage,${PHONES3VM})
> exten => 2003,3,Hangup
>
> ;
> ; IVR CHOICE 1
> ;
> exten => 1,1,Answer
> exten => 1,2,Playback(tt-somethingwrong)
> exten => 1,3,Playback(tt-monkeysintro)
> exten => 1,4,Playback(tt-monkeys)
> exten => 1,5,Hangup
>
> ;
> ; IVR RECORDER
> ;
> ; Record voice file to /tmp directory
> exten => 205,1,Wait(2) ; Call 205 to Record new Sound Files
> exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> exten => 205,3,Wait(2)
> exten => 205,4,Playback(/tmp/asterisk-recording)
> exten => 205,5,wait(2)
> exten => 205,6,Hangup
>
>
> ;
> ;MUSIC ON HOLD EXTENSION
> ;
> exten => 6000,1,Answer
> exten => 6000,2,SetMusicOnHold(default)
> exten => 6000,3,MusicOnHold()
> exten => 6000,4,Hangup
>
> ; ---------------------------------------------
> ; END DEFINE EXTENSIONS
> ; ----------------------------------------------
>
> ;------------------------------------
> ; RING EVERYONE EXTENSIONS
> ;-----------------------------------
>
> ;
> ; ring everyone
> ;
> 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,40000
>
> 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:JohnB@listbrokers.com]
> Sent: Tuesday, July 20, 2004 10:52 PM
> To: asterisk-users@lists.digium.com
> 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:johnb@listbrokers.com]
>>Sent: Tuesday, July 20, 2004 3:48 PM
>>To: asterisk-users@lists.digium.com
>>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 msg.bypassInstantMessage="1">
>>> <mwi msg.mwi.1.subscribe="8"
>>>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=""/>
>>> </msg>
>>> <nat nat.ip="" nat.signalPort=""
nat.mediaPortStart=""/>
>>> <user_preferences up.headsetMode="0"
up.useDirectoryNames="0"
>>>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:JohnB@listbrokers.com]
>>>Sent: Monday, July 19, 2004 10:17 PM
>>>To: asterisk-users@lists.digium.com
>>>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 msg.bypassInstantMessage="1">
>>><mwi msg.mwi.1.subscribe=""
msg.mwi.1.callBackMode="contact"
>>>msg.mwi.1.callBack="76" .... >
>>>
>>>3) In your extensions.conf, have something like:
>>>
>>>exten => 76,1,VoiceMailMain2(${EXTEN}@whatever_you_have_here)
>>>
>>>(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
>>>>Asterisk-Users@lists.digium.com
>>>>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
>>>Asterisk-Users@lists.digium.com
>>>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
>>>Asterisk-Users@lists.digium.com
>>>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
>>Asterisk-Users@lists.digium.com
>>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
>>Asterisk-Users@lists.digium.com
>>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
> Asterisk-Users@lists.digium.com
> 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
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
_______________________________________________
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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:JohnB@listbrokers.com]
Sent: Wednesday, July 21, 2004 7:04 PM
To: asterisk-users@lists.digium.com
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=XXXXXXXXX
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms
away
protocol=udp
dtmfmode=rfc2833
mailbox=7001@dont_spam_me
nat=0
disallow=all
allow=ulaw
allow=gsm
auth=md5
and the relevant line from voicemail.conf is
[listbrokers]
7001 => XXXXX,John Baker,JohnB@dont_spam_me.com,,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:${TRUNKMSD}})
> exten => _9011.,2,Congestion
>
> [trunkld]
> ;
> ; Long distance context accessed through trunk
> ;
> exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91NXXNXXXXXX,2,Congestion
>
> [trunklocal]
> ;
> ; Local seven-digit dialing accessed through trunk interface
> ;
> exten => _9480XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9480XXXXXXX,2,Congestion
>
> exten => _9602XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9602XXXXXXX,2,Congestion
>
> exten => _9623XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9623XXXXXXX,2,Congestion
>
>
> exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9NXXXXXX,2,Congestion
>
> [trunktollfree]
> ;
> ; Long distance context accessed through trunk interface
> ;
> exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91800NXXXXXX,2,Congestion
> exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXXXXXX,2,Congestion
> exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXXXXXX,2,Congestion
> exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXXXXXX,2,Congestion
>
> [international]
> ;
> ; Master context for international long distance
> ;
> ignorepat => 9
> include => longdistance
> include => trunkint
>
> [longdistance]
> ;
> ; Master context for long distance
> ;
> ignorepat => 9
> include => local
> include => trunkld
>
> [local]
> ;
> ; Master context for local, toll-free, and iaxtel calls only
> ;
> ignorepat => 9
> include => trunklocal
> include => trunktollfree
> ;
> ; You can use an alternative switch type as well, to resolve
> ; extensions that are not known here, for example with remote
> ; IAX switching you transparently get access to the remote
> ; Asterisk PBX
> ;
> ; switch => IAX2/user:password@bigserver/local
>
> ; now if we dial 8, we can check voicemail.
> ;
>
> ;exten => 8,1,VoicemailMain(s${CALLERIDNUM}) ;= no password
> exten => 8,1,VoicemailMain(${CALLERIDNUM}) ;= password
> exten => 8,2,Hangup
>
> ; Add some more extensions for the two extensions . now we'll be able
to> call one extension from the other.
> ; And if no one answers, it will go to the mailbox for that extension.
> ;
> ; extension 2000
> ;
> exten => 2000,1,Dial(${PHONES1},20,trf)
> exten => 2000,2,Macro(vmessage,${PHONES0VM})
> exten => 2000,3,Hangup
> ;
> ; extension 2001
> ;
> exten => 2001,1,Dial(${PHONES1},20,trf)
> exten => 2001,2,Macro(vmessage,${PHONES1VM})
> exten => 2001,3,Hangup
> ;
> ; extension 2002
> ;
> exten => 2002,1,Dial(${PHONES2},20,trf)
> exten => 2002,2,Macro(vmessage,${PHONES2VM})
> exten => 2002,3,Hangup
> ;
> ; extension 2003
> ;
> exten => 2003,1,Dial(${PHONES3},20,trf)
> exten => 2003,2,Macro(vmessage,${PHONES3VM})
> exten => 2003,3,Hangup
>
> ;
> ; IVR CHOICE 1
> ;
> exten => 1,1,Answer
> exten => 1,2,Playback(tt-somethingwrong)
> exten => 1,3,Playback(tt-monkeysintro)
> exten => 1,4,Playback(tt-monkeys)
> exten => 1,5,Hangup
>
> ;
> ; IVR RECORDER
> ;
> ; Record voice file to /tmp directory
> exten => 205,1,Wait(2) ; Call 205 to Record new Sound Files
> exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> exten => 205,3,Wait(2)
> exten => 205,4,Playback(/tmp/asterisk-recording)
> exten => 205,5,wait(2)
> exten => 205,6,Hangup
>
>
> ;
> ;MUSIC ON HOLD EXTENSION
> ;
> exten => 6000,1,Answer
> exten => 6000,2,SetMusicOnHold(default)
> exten => 6000,3,MusicOnHold()
> exten => 6000,4,Hangup
>
> ; ---------------------------------------------
> ; END DEFINE EXTENSIONS
> ; ----------------------------------------------
>
> ;------------------------------------
> ; RING EVERYONE EXTENSIONS
> ;-----------------------------------
>
> ;
> ; ring everyone
> ;
> 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,40000
>
> 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:JohnB@listbrokers.com]
> Sent: Tuesday, July 20, 2004 10:52 PM
> To: asterisk-users@lists.digium.com
> 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:johnb@listbrokers.com]
>>Sent: Tuesday, July 20, 2004 3:48 PM
>>To: asterisk-users@lists.digium.com
>>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 msg.bypassInstantMessage="1">
>>> <mwi msg.mwi.1.subscribe="8"
>>>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=""/>
>>> </msg>
>>> <nat nat.ip="" nat.signalPort=""
nat.mediaPortStart=""/>
>>> <user_preferences up.headsetMode="0"
up.useDirectoryNames="0"
>>>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:JohnB@listbrokers.com]
>>>Sent: Monday, July 19, 2004 10:17 PM
>>>To: asterisk-users@lists.digium.com
>>>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 msg.bypassInstantMessage="1">
>>><mwi msg.mwi.1.subscribe=""
msg.mwi.1.callBackMode="contact"
>>>msg.mwi.1.callBack="76" .... >
>>>
>>>3) In your extensions.conf, have something like:
>>>
>>>exten => 76,1,VoiceMailMain2(${EXTEN}@whatever_you_have_here)
>>>
>>>(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
>>>>
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>>>>
>>>>
>>>
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>>>
>>
>>
>>
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>>
>
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Hi, Has anyone gotten a D-Link DPH-80S SIP hard phone working with *? I should be receiving one of these in the next day or two and hoped to get a head-start on sip.conf additions for it. TIA. Michael Swan Neon Software, Inc.
where can you buy these? On Thu, 22 Jul 2004 10:12:59 -0700, Michael Swan <swan@neon.com> wrote:> Hi, > > Has anyone gotten a D-Link DPH-80S SIP hard phone working with *? I should > be receiving one of these in the next day or two and hoped to get a head-start > on sip.conf additions for it. TIA. > > Michael Swan > Neon Software, Inc. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
<-SNIP-> where can you buy these? <-SNIP-> Step 1. Open your web browser. Step 2. Access www.google.com Step 3. Paste "D-Link DPH-80S SIP" Hit the Google Search button Step 4. Following links Viola. The internet is a wonderful, wonderful thing. Alternatively for the *lazy* (ie: you) http://www.dlink.com/sales/where2buy/