Re: [Asterisk-Users] 3 Phone Call Qualtiy Issues
Hi, Thanks for the reply... The clicks are on every call and every few minutes or so, I guess you could call it regular intermittance? :) The only option for DTMF on my IAX phones are inband and outband-neither of which will respond to a menu prompt. I discovered I had to use outband just to use the phone's voicemail menu, so that is what it is set to. I cant specify DTMF in the IAX extension setup either- only on SIP extensions, for which I have no devices. For clarification on the outgoing dialing- I dial a number, the phone says Please Dial... for approx 9 secs, and then says Calling #. I have looked at all the places I can think of for max silence settings (thinking it would be like the voicemail maxsilence=# variable) but no luck. R Martin Joseph wrote: On Dec 20, 2005, at 1:01 PM, Rhonda Herron wrote: Hi, I have been battling with the following problems for a while and was hoping someone could shed some light on the subject. I am using AT320 402 IAX2 phones with 1.49 firmware (latest) connected to an Asterisk server running [EMAIL PROTECTED] 2.1 (includes Asterisk 1.2 and AMP 1.10). I am using ulaw codec but will upgrading to g729 within the week. And on to the questions: G.729 definitely helped my call quality across the internet. Within my LAN ulaw works great and sounds better. 1. During all calls there are periodic 'clicks' that sound as if one of the party's have hung up, about every 2 minutes or so. hmmm, Is this intermittent? 2. When calling, lets say for example Dell tech support, if the user is prompted to select a menu option the number they dial is not translated and the menu just keeps on going. I couldn't use inband DTMF with my equipment, I had to use rfc2833, which is set in both my equipment and the extensions that correspond. 3. The time between dialing an outgoing number and actual connection is painfully long- is this configurable in Asterisk somewhere? You might looks for extra waits in your dial plan extension that calls out? I don't notice mine being too long, although I hear two different dial tones at times. Depends on how your calls are terminated. Oh, and my outgoing provider is voipjet and incoming is Teliax. Teliax has been out for two days here for me, although it's back now... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rhonda Herron Network Administrator Reliance Software Systems, Inc. 30500 Northwestern Highway, Suite 415 Farmington Hills, MI 48334 Tel 248.865.7850 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 Phone Call Qualtiy Issues
Hi, I have been battling with the following problems for a while and was hoping someone could shed some light on the subject. I am using AT320 402 IAX2 phones with 1.49 firmware (latest) connected to an Asterisk server running [EMAIL PROTECTED] 2.1 (includes Asterisk 1.2 and AMP 1.10). I am using ulaw codec but will upgrading to g729 within the week. And on to the questions: 1. During all calls there are periodic 'clicks' that sound as if one of the party's have hung up, about every 2 minutes or so. 2. When calling, lets say for example Dell tech support, if the user is prompted to select a menu option the number they dial is not translated and the menu just keeps on going. 3. The time between dialing an outgoing number and actual connection is painfully long- is this configurable in Asterisk somewhere? Oh, and my outgoing provider is voipjet and incoming is Teliax. Thanks! -R ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer
Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Call Transfer
I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Call Transfer
It is set to rfc2833. Tom Vile wrote: maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Call Transfer
Yes, I can dial *97 for VM and check messages. When I select # during a call it does nothing though. I tried inband for DTMF but that didnt work. Am going to run debug mode ( first I have to figure out how :) ) and I will let you know what I find out. Thanks so far, R Tom Vile wrote: Blind transfer should work fine #. Can you dial into Voicemail and enter your password succesfully? On 10/20/05, *BJ Weschke* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It is set to rfc2833. Tom Vile wrote: maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com
Re: [Asterisk-Users] Call Transfer
He is what happens from the time the extension is selected from the time
the digital receptionist answers until I hangup. I watched the logs as
I was pushing all sorts of transfer button possibilities and nothing. It
just stayed at 'ooh, voice format changed to 4' Which, while humorous
tells me nothing except that my phone is not able to communicate with
the sever at all from the time the call is put through until the call is
done.
Oct 20 15:33:45 VERBOSE[2909]: -- Executing
Dial(IAX2/[EMAIL PROTECTED]/4, IAX2/7878|15|tr) in new stack
Oct 20 15:33:45 DEBUG[2909]: SIMPLE DIAL (NO URL)
Oct 20 15:33:45 VERBOSE[2909]: -- Called 7878
Oct 20 15:33:45 VERBOSE[2909]: -- Call accepted by xxx.xxx.xxx.xxx
(format ulaw)
Oct 20 15:33:45 VERBOSE[2909]: -- Format for call is ulaw
Oct 20 15:33:45 VERBOSE[2909]: -- IAX2/7878/8 is ringing
Oct 20 15:33:50 VERBOSE[2909]: -- IAX2/7878/8 answered
IAX2/[EMAIL PROTECTED]/4
Oct 20 15:33:50 VERBOSE[2909]: -- Attempting native bridge of
IAX2/[EMAIL PROTECTED]/4 and IAX2/7878/8
Oct 20 15:33:50 DEBUG[2909]: Ooh, voice format changed to 4
Here is the extension config for 7878:
exten = 7878,1,Macro(exten-vm,[EMAIL PROTECTED],7878)
And this is the config for aah_1 ( our digital receptionist)
[aa_1]
include = aa_1-custom
exten = 1,1,Goto(,s,1);
exten = fax,1,Goto(ext-fax,in_fax,1);
exten = h,1,Hangup();
exten = i,1,Playback(invalid);
exten = i,2,Goto(s,7);
include = ext-local
include = app-messagecenter
include = app-directory
exten = s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4);
exten = s,2,Answer();
exten = s,3,Wait(1);
exten = s,4,SetVar(DIR-CONTEXT=default);
exten = s,5,DigitTimeout(3); Basic
exten = s,6,ResponseTimeout(7);
exten = s,7,Background(custom/aa_1);
Thanks, once again,
-R
Rhonda Herron wrote:
Yes, I can dial *97 for VM and check messages. When I select # during
a call it does nothing though. I tried inband for DTMF but that didnt
work. Am going to run debug mode ( first I have to figure out how :) )
and I will let you know what I find out.
Thanks so far,
R
Tom Vile wrote:
Blind transfer should work fine #. Can you dial into Voicemail and
enter your password succesfully?
On 10/20/05, *BJ Weschke* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I'm not sure the txfer functionality is in the 1.0.X branch. I'm
pretty sure you will need HEAD or the 1.2 betas.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
It is set to rfc2833.
Tom Vile wrote:
maybe its not setting the DTMF tones properly. What do you
have setup
for the phone and extensions. Usually its rfc2833 but could
be inband.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I have the phone specific directions to transfer calls,
but I
tried your
suggestion. No go. I have 3 of the Eezee phones and call
transfer
doesn't work on any of them, so I really don't think it
is hardware
related. I think the problem may be with my feature.conf
which had no
reference to blindxfer or atxfer. I added them so my
feature.conf now
looks like this:
transferdigittimeout = 3 ; Number of seconds to
wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to
the parked
caller
; when someone dials a
parked call
xfersound = beep ; to indicate an attended
transfer is
complete
xferfailsound = beeperr; to indicate a failed
transfer
;adsipark = yes ; if you want ADSI parking
announcements
;pickupexten = *8 ; Configure the pickup
extension.
Default is *8
;featuredigittimeout = 500 ; Max time (ms) between
digits for
; feature
activation. Default is 500
[featuremap]
blindxfer = #; Blind transfer
disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = *2
I rebooted my * server but still no go. Are
there dependencies I am
not aware of? Should [featuremap] be referenced elsewhere
as well?
I am
working with * CVS 1.0.9 and have found an article on
wiki that
support
for call transfer was added in 1.2. Are there other
places I need to
hack for this functionality?
Thanks,
-R
Tom Vile wrote:
try # and then dial the extension.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED
