Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

On Wed, Jun 6, 2018 at 1:51 AM Olivier wrote: > > > 2018-06-05 20:29 GMT+02:00 George Joseph : > >> >> >> On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote: >> >>> >>> >>> 2018-06-05 15:27 GMT+02:00 George Joseph : >>> Thank you very much, George for replying. >>> On Tue, Jun 5, 2018

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

2018-06-05 20:29 GMT+02:00 George Joseph : > > > On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote: > >> >> >> 2018-06-05 15:27 GMT+02:00 George Joseph : >> Thank you very much, George for replying. >> >>> >>> >>> On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote: >>> Hi, After a long dis

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote: > > > 2018-06-05 15:27 GMT+02:00 George Joseph : > Thank you very much, George for replying. > >> >> >> On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote: >> >>> Hi, >>> >>> After a long discussion with a friend, I would like to ask here: >>> >>> 1.Acco

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

2018-06-05 15:27 GMT+02:00 George Joseph : Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote: > Hi, > > After a long discussion with a friend, I would like to ask here: > > 1.According SIP RFCs, is possible/recommended to have different values in > From and P-Asserted-Id fields ? > For instance, From field showing 123456789 and P-Asserted-Id s

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

On Tue, Jun 05, 2018 at 11:34:51AM +0200, Olivier wrote: > 1.According SIP RFCs, is possible/recommended to have different values in > From and P-Asserted-Id fields ? > For instance, From field showing 123456789 and P-Asserted-Id showing > 987654321 (beside privacy considerations) ? Yes, most obvi

[asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When

Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

On 29/03/17 16:18, Olivier wrote: Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complie

Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

On Wed, Mar 29, 2017 at 06:04:49PM +0200, Olivier wrote: > Is there any relation between this external patch and the binary mentioned > in [2] > [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/ > > The later one mentions a binary-only distribution to comply with legal > constraints. No,

Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

2017-03-29 17:28 GMT+02:00 Tzafrir Cohen : > On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote: > > Hello, > > > > After reading [1] (in french), I would be very happy if I could get > answers > > to: > > > > 1. Does this 13.7+20161113-3 package version has any relation with > > asterisk's v

Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote: > Hello, > > After reading [1] (in french), I would be very happy if I could get answers > to: > > 1. Does this 13.7+20161113-3 package version has any relation with > asterisk's version it complements ? Current asterisk version in repo is

[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complies with it ? 2. From package description

[asterisk-users] Questions regarding Dial's D option

Hello, I'm currently playing with Application Dial D option. This option is documented with: D([called][:calling[:progress]]): Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The DTMF string is sent to the called party, and th

Re: [asterisk-users] Questions... connecting Asterisk to the World

On Saturday 14 May 2016, Stefan Becker wrote: > Greetings, > > asterisk list and community, > > I have a problem in how our telefon switch (Siemens HiCOM) > "talks" with my new configured Asterisk server (V.11.18.0) > > without my Asterisks server in the middle > > <--> Siemens HiCOM <-ISD

Re: [asterisk-users] Questions... connecting Asterisk to the World

On Sat, 14 May 2016, Stefan Becker wrote: On Sat, 14 May 2016, Steve Edwards wrote: I think you need to make the outbound dial a single 'transaction' either by using an extension pattern that includes the 0 like '055' to dial 555-555- or eliminate the 0 (and the idiom of 'request

Re: [asterisk-users] Questions... connecting Asterisk to the World

On Sat, 14 May 2016, Steve Edwards wrote: > I think you need to make the outbound dial a single 'transaction' either by > using an extension pattern that includes the 0 like '055' to dial > 555-555- or eliminate the 0 (and the idiom of 'requesting an outgoing > line') and detect an in

Re: [asterisk-users] Questions... connecting Asterisk to the World

On Sat, 14 May 2016, Stefan Becker wrote: A phone connected to the switch requests an "Outgoing" line by dialing "0". --> Asterisks recieves incoming call on "s". The dialed digits are collected. The dial plan is executed accordingly but the "caller" recieves no more information about the dial

[asterisk-users] Questions... connecting Asterisk to the World

Greetings, asterisk list and community, I have a problem in how our telefon switch (Siemens HiCOM) "talks" with my new configured Asterisk server (V.11.18.0) without my Asterisks server in the middle <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom A phone connected to the switch req

Re: [asterisk-users] Questions regarding ICE and STUN with Asterisk

Well, after a more specific research I came to 2 conclusions: 1) no need to specify "stunaddr" option in Asterisk configs in this case, as we know that host definitely has a public IP 2) as of the inclusion of all local IP-addresses as candidates, this is (apparently) done in "rtp_add_candida

[asterisk-users] Questions regarding ICE and STUN with Asterisk

Hi everyone I would like to get some help and clarification from the experienced ones, upon the following: - we're using Asterisk 13.7.0, that is deployed on a host, that has a public IP *and* a couple of gray IPs (192.168.x.x & 10.10.x.x) - we're using WebRTC web-page (jsSIP) as a client

[asterisk-users] Questions about API Asterisk Java

Hello everyone, I have some questions about one of the Asterisk API called Asterisk Java: * How did it work? * How can we use it in order to connect one external program to Asterisk? * Can we use it with Asterisk 12? Thank you in advance, Pierre -- Maybe, I have alr

Re: [asterisk-users] Questions on musiconhold.conf custom mode

2014-10-27 11:50 GMT+01:00 Thorsten Göllner : > > Am 27.10.2014 08:54, schrieb Olivier: >> 2014-10-25 19:33 GMT+02:00 Thorsten Göllner : >>> Am 25.10.2014 00:09, schrieb Olivier: Hello, I need to play some musiconhold content starting at a random duration from the start. >>

Re: [asterisk-users] Questions on musiconhold.conf custom mode

Am 27.10.2014 08:54, schrieb Olivier: > 2014-10-25 19:33 GMT+02:00 Thorsten Göllner : >> Am 25.10.2014 00:09, schrieb Olivier: >>> Hello, >>> >>> I need to play some musiconhold content starting at a random duration >>> from the start. >>> >>> Thanks to mode=custom option and either madplay or mpg

Re: [asterisk-users] Questions on musiconhold.conf custom mode

2014-10-25 19:33 GMT+02:00 Thorsten Göllner : > > Am 25.10.2014 00:09, schrieb Olivier: >> Hello, >> >> I need to play some musiconhold content starting at a random duration >> from the start. >> >> Thanks to mode=custom option and either madplay or mpg123 programs, I >> could successfully get what

Re: [asterisk-users] Questions on musiconhold.conf custom mode

Am 25.10.2014 00:09, schrieb Olivier: > Hello, > > I need to play some musiconhold content starting at a random duration > from the start. > > Thanks to mode=custom option and either madplay or mpg123 programs, I > could successfully get what I was after on a Debian Wheezy system. > > Now I realiz

[asterisk-users] Questions on musiconhold.conf custom mode

Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) ca

Re: [asterisk-users] Questions about chan_dahdi, PRI, MWI (and Q.SIG)

On Fri, Jun 28, 2013 at 3:59 AM, Jens Bürger wrote: > Hello everyone, > > My setup: > Debian squeeze > Asterisk 1.8, DAHDI, libpri, compiled from source > TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 > > I'm trying to get MWI for Voicemail working. In the same server I have > a

[asterisk-users] Questions about chan_dahdi, PRI, MWI (and Q.SIG)

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrate

Re: [asterisk-users] Questions about sRTP

On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl wrote: > > > On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp wrote: > >> Mike Diehl wrote: >> >>> Hi all, >>> >>> I'm getting ready to setup SIP/TLS and SRTP. But I have a few >>> questions. The first one is that I was reading an article at: >>> >>> htt

Re: [asterisk-users] Questions about sRTP

On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp wrote: > Mike Diehl wrote: > >> Hi all, >> >> I'm getting ready to setup SIP/TLS and SRTP. But I have a few >> questions. The first one is that I was reading an article at: >> >> https://supportforums.cisco.com/docs/DOC-15381 >> >> That indicated tha

Re: [asterisk-users] Questions about sRTP

Mike Diehl wrote: Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or

[asterisk-users] Questions about sRTP

Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically

Re: [asterisk-users] Questions about extension.conf

Excellent. It appears that Getting Started has a lot more stuff in it than the documentation for 1.8. Very helpful. Ron On 29/11/2012 12:31 PM, David M. Lee wrote: On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote: That is a good answer. Thanks. Any reason why it is not documented? It's d

Re: [asterisk-users] Questions about extension.conf

On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote: > That is a good answer. > Thanks. > Any reason why it is not documented? It's documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities > Ron -- David M. Lee Digium, Inc. | Software Dev

Re: [asterisk-users] Questions about extension.conf

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Thursday, November 29, 2012 12:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Questions about extension.conf On 29/11/2012 11:47 AM, Salman Zafar wrote: It is self explanatory, for example

Re: [asterisk-users] Questions about extension.conf

That is a good answer. Thanks. Any reason why it is not documented? Ron On 29/11/2012 11:52 AM, Mikhail Lischuk wrote: Shitian Long wrote 29.11.2012 18:40: There is a part of dial plan from sample extension.conf above. My Question is how "same =>" key word works . Thanks "same" is use

Re: [asterisk-users] Questions about extension.conf

On 29/11/2012 11:47 AM, Salman Zafar wrote: It is self explanatory, for example: exten => _X.,1, Noop("Let say we have allowed all numbers i.e. _X means and . specifies any range") same => n,NoOp("Here we have skipped mentioning dial-pattern again and thats it") Hope I have answered your q

Re: [asterisk-users] Questions about extension.conf

Shitian Long wrote 29.11.2012 18:40: > There is a part of dial plan from sample extension.conf above. My Question is how "same =>" key word works . > > Thanks "same" is used for complex templates, if you don't want to copy previous line or afraid you can make a typo. exten => _1XXNXXX,1,A

Re: [asterisk-users] Questions about extension.conf

ers-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long Sent: Thursday, November 29, 2012 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Questions about extension.conf Hello I have been reading th

[asterisk-users] Questions about extension.conf

Hello I have been reading the sample extension.conf ;### [outbound-freenum2] ; This is the handler which performs the dialing logic. It is called ; from the [outbound-freenum] context ; exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN}) same => n,Set(SUFFIX=${CUT(

Re: [asterisk-users] Questions on converting to ConfBridge

On 02/10/12 06:07 PM, Richard Kenner wrote: I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. > There also doesn't seem to be a way to lock conferences or mute or kick out u

Re: [asterisk-users] Questions on converting to ConfBridge

> Why are you wanting to use CLI commands instead of AMI? The available > AMI actions for ConfBridge can do listing/locking/muting/kicking etc as > you want. Because I can't easily manually do an AMI command, but instead have to write code to do it. It's important to me to be able to clean up t

Re: [asterisk-users] Questions on converting to ConfBridge

Hola, Richard Kenner wrote: I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked ar

[asterisk-users] Questions on converting to ConfBridge

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the

[asterisk-users] Questions about fax detection

Hello, I want to offer SIP phone user a "custom fax-to-email" feature. Here is how I would describe this feature: - for every SIP phone,a custom email address is defined - when a SIP phone answers an incoming call (from a trunk or another SIP endpoint), it detects the call is coming from a fax ma

Re: [asterisk-users] Questions on hardware or software-based echo cancellation

On 01/13/2012 02:12 AM, Olivier wrote: 2012/1/12, Kevin P. Fleming: On 01/12/2012 06:39 AM, Olivier wrote: Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ) for cases when a hardware ech canceller is present

Re: [asterisk-users] Questions on hardware or software-based echo cancellation

2012/1/12, Kevin P. Fleming : > On 01/12/2012 06:39 AM, Olivier wrote: >> Hi, >> >> I'm having some questions related to echo cancellation configuration >> on a Digium board enabled systems (B410P, TE420, TE420B, ) for >> cases when a hardware ech canceller is present or not. >> >> I read in TE

Re: [asterisk-users] Questions on hardware or software-based echo cancellation

On 01/12/2012 06:39 AM, Olivier wrote: Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in ch

[asterisk-users] Questions on hardware or software-based echo cancellation

Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in chan_dahdi.conf on a VPMOCT64-equiped syste

Re: [asterisk-users] Questions on IAX client

On Sun, 23 Oct 2011, asterisk asterisk wrote: I used to use Zoiper IAX to connect to my asterisk server from remote site. On asterisk CLI, I can see my zoiper client registered and stay on line. HOwever, I don't know why now I can't call this client. It always show up as "Unable to create chan

[asterisk-users] Questions on IAX client

Hi, I used to use Zoiper IAX to connect to my asterisk server from remote site. On asterisk CLI, I can see my zoiper client registered and stay on line. HOwever, I don't know why now I can't call this client. It always show up as "Unable to create channel IAX2 (Cause 20 Unknown) I am using Asteri

Re: [asterisk-users] Questions on Dahdi

On 10/05/2011 07:28 PM, asterisk asterisk wrote: I have naive question. I do not have any hardware on my asterisk host. All I have are either SIP trunk for DID or hardware ATA which bridges the asterisk to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I encounter problem in t

[asterisk-users] Questions on Dahdi

I have naive question. I do not have any hardware on my asterisk host. All I have are either SIP trunk for DID or hardware ATA which bridges the asterisk to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I encounter problem in this when I try to install Dahdi latest but I found

Re: [asterisk-users] Questions about FMFM with linked servers

: [asterisk-users] Questions about FMFM with linked servers All; In a linked server environment, running Asterisk 1.6 I am noticing that when a call is placed from server A to server B (via 4 digit extension) and server B ext has a FMFM to call their mobile, the mobile phone shows the default

[asterisk-users] Questions about FMFM with linked servers

All; In a linked server environment, running Asterisk 1.6 I am noticing that when a call is placed from server A to server B (via 4 digit extension) and server B ext has a FMFM to call their mobile, the mobile phone shows the default caller ID setting on server B instead of the actual caller id of

Re: [asterisk-users] Questions About Fax for Asterisk

On Sat, May 8, 2010 at 2:27 PM, David Backeberg wrote: > On Sat, May 8, 2010 at 7:21 AM, Steve Totaro > wrote: > > Maybe there is a simple setting somewhere, but "RTFM" from Digium tech > > support when the FM offers no suggestion on how to possibly tweak > settings > > for better success. > > Do

Re: [asterisk-users] Questions About Fax for Asterisk

On Sat, May 8, 2010 at 7:21 AM, Steve Totaro wrote: > Maybe there is a simple setting somewhere, but "RTFM" from Digium tech > support when the FM offers no suggestion on how to possibly tweak settings > for better success. Do you want to dump some samples from your dialplan? I know I personally

Re: [asterisk-users] Questions About Fax for Asterisk

On Fri, May 7, 2010 at 9:56 PM, Steve Underwood wrote: > On 05/08/2010 08:15 AM, Steve Totaro wrote: > > > > > > On Fri, May 7, 2010 at 2:01 PM, Martin > > wrote: > > > > On Thu, May 6, 2010 at 3:11 PM, Steve Totaro > > >

Re: [asterisk-users] Questions About Fax for Asterisk

On 05/08/2010 08:15 AM, Steve Totaro wrote: > > > On Fri, May 7, 2010 at 2:01 PM, Martin > wrote: > > On Thu, May 6, 2010 at 3:11 PM, Steve Totaro > > wrote: > > Yes, I purchased licenses for Fax for Asteris

Re: [asterisk-users] Questions About Fax for Asterisk

On Fri, May 7, 2010 at 2:01 PM, Martin wrote: > On Thu, May 6, 2010 at 3:11 PM, Steve Totaro > wrote: > > Yes, I purchased licenses for Fax for Asterisk and yes I called tech > support > > and had the WORST experience I have ever had with any technical support > > call. > > > > I am running Aste

Re: [asterisk-users] Questions About Fax for Asterisk

On Thu, May 6, 2010 at 3:11 PM, Steve Totaro wrote: > Yes, I purchased licenses for Fax for Asterisk and yes I called tech support > and had the WORST experience I have ever had with any technical support > call. > > I am running Asterisk 1.6.2.6 and: > > FAX For Asterisk Components: >     App

[asterisk-users] Questions About Fax for Asterisk

Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (

Re: [asterisk-users] Questions about asterisk and spa2102

On Fri, 29 Jan 2010, Kosa wrote: > 1.- is it possible to use an spa2102 to make and revice calls from a > "normal" phone? I mean, I know I can use it to connect an analog to an > asterisk server, but I want to know if it can be used to connect > asterisk to the analog phoneline. The 2102 is an

Re: [asterisk-users] Questions about asterisk and spa2102

Kosa wrote: > Hi there! First mail on the list :) > > 1.- is it possible to use an spa2102 to make and revice calls from a "normal" > phone? I mean, I know I can use it to connect an analog to an asterisk > server, but I want to know if it can be used to connect asterisk to the > analog phonel

[asterisk-users] Questions about asterisk and spa2102

Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unl

Re: [asterisk-users] Questions about static

> We swapped PoE switches, phones, cable and switch ports multiple times. > What do you mean by local interference? Cell phone? The person swears > nothing is near the phone. There are lots of things that can cause interference. Radios, elevators, bad electrical wiring, you name it. Is the stati

Re: [asterisk-users] Questions about static

Sent: Friday, November 27, 2009 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about static We swapped PoE switches, phones, cable and switch ports multiple times. What do you mean by local interference? Cell phone? The person swears nothi

Re: [asterisk-users] Questions about static

...@lists.digium.com] On Behalf Of Noah Miller Sent: Friday, November 27, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about static > We have swapped out the phone multiple times for the user. > Only one user. Bad PoE port on the

Re: [asterisk-users] Questions about static

> We have swapped out the phone multiple times for the user. > Only one user. Bad PoE port on the switch? How about local interference that the user cannot control? Does the same phone experience static when moved elsewhere? Do you have a power brick for the phone so you can try it as non-PoE?

Re: [asterisk-users] Questions about static

List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Questions about static Is it a single user? Or every single phone? If it’s a single user, and you can get hold of a UPS with power conditioning on it, try plugging the various devices into it – there might be some dirty power

Re: [asterisk-users] Questions about static

-Commercial Discussion Subject: Re: [asterisk-users] Questions about static On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote: > Would be a cause of static for inbound/outbound and ext to ext calls? > > Its voip both in and out. > > We swapped, phones, cordes, switches etc... > >

Re: [asterisk-users] Questions about static

On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote: > Would be a cause of static for inbound/outbound and ext to ext calls? > > Its voip both in and out. > > We swapped, phones, cordes, switches etc….. > > Typically a reboot of the phone resolves the problem…person also > swears there is nothing on

Re: [asterisk-users] Questions about static

-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Thursday, 26 November 2009 07:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Questions about static Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/out

[asterisk-users] Questions about static

Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc….. Typically a reboot of the phone resolves the problem…person also swears there is nothing on

Re: [asterisk-users] Questions about Voicemail

, November 25, 2009 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about Voicemail On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote: > Regarding the email to multiple receipients, it is available on an ad-hoc > basis from the

Re: [asterisk-users] Questions about Voicemail

On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote: > Regarding the email to multiple receipients, it is available on an ad-hoc > basis from the phone? > > IE; call into the voicemail system, enter x digit to send a voicemail to > multiple users, record the message, then enter the destination m

Re: [asterisk-users] Questions about Voicemail

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: Monday, November 23, 2009 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about Voicemail On Mon, 2009-11-23 at

Re: [asterisk-users] Questions about Voicemail

On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote: > I am sorry if this is not the correct place to post a question. > > I am wondering if there is way in asterisk 1.2 to: > > 1. Send a voicemail (from the phone) to multiple recipients. Yes I believe so. 1. The voicemail app allows deliv

[asterisk-users] Questions about Voicemail

I am sorry if this is not the correct place to post a question. I am wondering if there is way in asterisk 1.2 to: 1. Send a voicemail (from the phone) to multiple recipients. 2. Require (as an admin) for users 1st logging into their voicemail to change their greeting and/or pass

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

2009/11/12 Tzafrir Cohen > On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote: > > 2009/11/11 Tzafrir Cohen > > > > > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > > > > > What about adding per-span section headers like Asterisk .conf files > ? > > > > [span1] > > > > group_lin

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote: > 2009/11/11 Tzafrir Cohen > > > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > > > What about adding per-span section headers like Asterisk .conf files ? > > > [span1] > > > group_lines 1 > > > pri_termtype > > >

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

2009/11/11 Tzafrir Cohen > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > > 2009/11/10 Tzafrir Cohen > > > > > On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: > > > > Hello, > > > > > > > > 1. How can specify in /etc/dahdi/genconf_parameters file that a port > from > > > a >

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > 2009/11/10 Tzafrir Cohen > > > On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: > > > Hello, > > > > > > 1. How can specify in /etc/dahdi/genconf_parameters file that a port from > > a > > > B410P board is to be "disabled". > > >

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

2009/11/10 Tzafrir Cohen > On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: > > Hello, > > > > 1. How can specify in /etc/dahdi/genconf_parameters file that a port from > a > > B410P board is to be "disabled". > > There's currently no way to do that. > > It should be trivial to implment.

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: > Hello, > > 1. How can specify in /etc/dahdi/genconf_parameters file that a port from a > B410P board is to be "disabled". There's currently no way to do that. It should be trivial to implment. The more difficult part of it would be how t

[asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

Hello, 1. How can specify in /etc/dahdi/genconf_parameters file that a port from a B410P board is to be "disabled". Playing with comments (see bellow) doesn't help : file /etc/asterisk/dahdi-channels.conf is filled with 4 ports data. pri_termtype SPAN/1 TE SPAN/2

[asterisk-users] Questions about app_jack.c [solved]

Hello, I corrected a bug and did some little optimizations in app_jack.c. It works great now. I propose this new file based on revision 140568. Fabien /* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2007 - 2008, Russell Bryant * * Russell Bryant * * See http://www.a

[asterisk-users] Questions about app_jack.c

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchanne

Re: [asterisk-users] Questions on X100P/X101P cards

>BTW, can someone explain to a libart major like me (;-)) where echo comes on in a telephone conversation? I seem to recall it's due to the length of the line between the CO and the local party, but I'm not sure. Yes, I'll tackle that. It takes a finite amount of time for the electrical signal o

Re: [asterisk-users] Questions on X100P/X101P cards

On Thu, 7 May 2009 13:40:20 +0300, Tzafrir Cohen wrote: >Another thing: their "global-line-standard" should basically (if >properly written) resolve http://bugs.digium.com/view.php?id=11057 . >Though I guess the new code will actually be in DAHDI, as Zaptel is >frozen. Ah yes, I seem to remember

Re: [asterisk-users] Questions on X100P/X101P cards

On Thu, 07 May 2009 10:16:55 -0400, Jon Pounder wrote: >yeah I agree with the above - I never really found echo to ever be a >problem, my only complaint was on some less than stellar cpu's I was >having dtmf recognition problems. BTW, can someone explain to a libart major like me (;-)) where ec

Re: [asterisk-users] Questions on X100P/X101P cards

Jonathan Moore wrote: > On Wed, May 6, 2009 at 10:53 PM, John Novack > wrote: > >> Not sure how you would do that, as the X100 card is an FXO card, won't >> provide either battery or dial tone to the cordless. >> What you will want for that is an FXS card or ATA. >> The X100 card will connect t

Re: [asterisk-users] Questions on X100P/X101P cards

On Wed, May 6, 2009 at 10:53 PM, John Novack wrote: > Not sure how you would do that, as the X100 card is an FXO card, won't > provide either battery or dial tone to the cordless. > What you will want for that is an FXS card or ATA. > The X100 card will connect to a central office line, and with t

Re: [asterisk-users] Questions on X100P/X101P cards

On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote: > X100P SE Setup Guide - Global Line Standards > http://novavox.co.uk/support/x100p.html > Richard Spencer supp...@novavox.co.uk Another thing: their "global-line-standard" should basically (if properly written) resolve http://bugs.digium.c

Re: [asterisk-users] Questions on X100P/X101P cards

On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote: > On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen > wrote: > >Some X100P cards (e.g.: those that are based on SI3034, but not those > >basedon SI3035) support "programmable" impedance settings. Sadly the > >wcfxo driver does not support it.

Re: [asterisk-users] Questions on X100P/X101P cards

On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen wrote: >Some X100P cards (e.g.: those that are based on SI3034, but not those >basedon SI3035) support "programmable" impedance settings. Sadly the >wcfxo driver does not support it. > >Fixing it should mostly be a matter of lifting some code from w

Re: [asterisk-users] Questions on X100P/X101P cards

On Thu, May 07, 2009 at 07:24:21AM +0200, Massimo Nuvoli wrote: > John Novack ha scritto: > > > > > Not sure how you would do that, as the X100 card is an FXO card, > > won't provide either battery or dial tone to the cordless. What you > > will want for that is an FXS card or ATA. The X100 card

Re: [asterisk-users] Questions on X100P/X101P cards

John Novack ha scritto: > > Not sure how you would do that, as the X100 card is an FXO card, > won't provide either battery or dial tone to the cordless. What you > will want for that is an FXS card or ATA. The X100 card will > connect to a central office line, and with the later software echo >

Re: [asterisk-users] Questions on X100P/X101P cards

Jonathan Moore wrote: > On Wed, May 6, 2009 at 8:47 PM, ContactTel Business > wrote: > >> I'd say in life you get what you pay for.. and sometime you even pay for >> stuff that should be free.. >> > > I have to agree. > > I have a few of these cards I started out with. They were great f

Re: [asterisk-users] Questions on X100P/X101P cards

On Wed, May 6, 2009 at 8:47 PM, ContactTel Business wrote: > I'd say in life you get what you pay for.. and sometime you even pay for > stuff that should be free.. I have to agree. I have a few of these cards I started out with. They were great for the "wow, I finally got asterisk to do somethi

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