Re: [asterisk-users] how to route outgoing calls on IP-level

Hi, I would recommend instead of Using IPs in your Billing, your Prefixes. Most of the billing softwares can to billing based on Prefix, for example when "Bill Clinton" from Extension 100 is calling, add 22 or 22# in front of the calling number 22#12345678, then your billing can do the rest based

Re: [asterisk-users] how to route outgoing calls on IP-level

there's just one factor - customer, i.e. extension in terms of Asterisk. On 9/15/07, Joseph Bajin <[EMAIL PROTECTED]> wrote: > > What are the factors in deciding which interface the traffic needs to > go out of? > > Is it based on IP address, is it based on the terminating carrier? > > --Joe > > O

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On Friday 14 September 2007 04:12:48 pm Atis wrote: > exten => _XXX*.,1,Goto(default-wildcard|${EXTEN}|1) > exten => _*.,1,Goto(default-wildcard|${EXTEN}|1) > exten => _X*.,1,Goto(default-wildcard|${EXTEN}|1) excellent sir! thank you! actually, since i'm using this for testing ISN/ITAD,

Re: [asterisk-users] how to route outgoing calls on IP-level

What are the factors in deciding which interface the traffic needs to go out of? Is it based on IP address, is it based on the terminating carrier? --Joe On 9/14/07, Kate Kretz <[EMAIL PROTECTED]> wrote: > Dear Sirs, > > out asterisk server has multiple network cards. > > I want some outgoing ca

Re: [asterisk-users] how to route outgoing calls on IP-level

well, the situation is: we have two-state VoIP-routing customers (h323,sip) ---> asterisk --> our home made h323 proxy the final billing is done at h323 proxy, and it distingushes customers by their IP addresses. so, if I want to bill two group of SIP customers separately, I need to route call

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

Jared Smith wrote: > On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: > >> . matches any number of the preceding character, change it to _X.*X. >> > > That still won't help. Once the Asterisk pattern matching parser sees a > period in the pattern, it ignores anything after it. (I

Re: [asterisk-users] CallWithUs Service?

In VOIP, your quality of your voice is as good as your network. if you want clear call quality, QOS is a must. Well, when the call leaves your network and enters internet, QOS is not enforced. As a general rule choose the closest to your network. for me its Teliax, i get to their proxy after 7 hops

Re: [asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400

i did have same issue with DISA in 1.4 and TDM400 FXO, I switched back to Authenticate and waitexten. On 9/14/07, Benjamin M. <[EMAIL PROTECTED]> wrote: > > > > Originally posted at http://forums.digium.co

Re: [asterisk-users] DECT SIP phones

On Fri, 2007-09-14 at 12:00 -0500, [EMAIL PROTECTED] wrote: > Date: Fri, 14 Sep 2007 09:32:35 -0500 > From: Tilghman Lesher <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] DECT SIP phones > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]>

Re: [asterisk-users] ztdummy kills audio

I'm seeing the problem on both etch and lenny releases. Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux > What release of Debian is it? > What kernel do you use? Packaged or self-built? > uname -a > ___ Sign up now for AstriC

Re: [asterisk-users] Force a new user to configure Comedian mail?

Russell Bryant wrote: > Doug Lytle wrote: > > I'm ... sorry? However, it does behave exactly as is documented. It > specifies > > No need to be, I was just making an observation. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Saf

Re: [asterisk-users] Force a new user to configure Comedian mail?

Doug Lytle wrote: > Russell Bryant wrote: >> >> Yep. In fact, it was one of the first patches I ever wrote for Asterisk. :) >> > > And under 1.2 it can be easily bypassed. After the password is changed, > if the user hangs up, the next time they call into the voice mail > system, it doesn'

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On Sat, 2007-09-15 at 00:12 +0300, Atis wrote: > On 9/14/07, Jared Smith <[EMAIL PROTECTED]> wrote: > > On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: > > > . matches any number of the preceding character, change it to _X.*X. > > > > That still won't help. Once the Asterisk pattern matc

Re: [asterisk-users] Force a new user to configure Comedian mail?

On Fri, Sep 14, 2007 at 05:04:09PM -0400, Dave Fullerton wrote: > Jeremy Wadhams wrote: > > In Asterisk 1.4, is there any way to "force" new users to configure > > their mailbox? I'm thinking a simple IVR that holds a user's hand > > through changing their PIN, recording their name, and setting up

Re: [asterisk-users] ztdummy kills audio

On Fri, Sep 14, 2007 at 04:41:22PM -0400, John Albano wrote: > Yes, that was after approx a minute. Output from lsmod is... > > zaptel182948 4 zttranscode,ztdummy > crc_ccitt 3072 1 zaptel What release of Debian is it? What kernel do you use? Packaged or self-bui

[asterisk-users] Zaptel ztdummy module causes playback to fail

I'm using asterisk 1.4.11 and Zaptel version 1.4.5.1 with kernel 2.6.22. I have the ztdummy module loaded, which is using zaptel and rtc. When the ztdummy module is loaded, sounds are not heard when using the asterisk "background" command. When the ztdummy module is unloaded, which also removes zap

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On Friday 14 September 2007 15:35:47 Anthony Messina wrote: > On Friday 14 September 2007 12:37:11 pm Tilghman Lesher wrote: > > On Friday 14 September 2007 11:39:40 Anthony Messina wrote: > > > I am working on getting freenum.org ISN/ITAD numbers integrated into my > > > exiting dialplan however I

Re: [asterisk-users] Force a new user to configure Comedian mail?

Russell Bryant wrote: > Jeremy Wadhams wrote: > > Yep. In fact, it was one of the first patches I ever wrote for Asterisk. :) > > And under 1.2 it can be easily bypassed. After the password is changed, if the user hangs up, the next time they call into the voice mail system, it doesn't

Re: [asterisk-users] AGI script fails on IAX channels (from call file).

Jonas Arndt wrote: > Hi Guys, > > I have already tried this one on the developers list. I have not been > successful getting much back there and I have notified them that I will > post this on the users list instead. Hopefully somebody have tried > something similar and can help out. > > I am devel

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

shadowym wrote: > Yes thank you for reminding me it is open source. Thank you for reminding > me people can write their own code for it. > > I'll get right on rewriting the entire sip code. Should only take me a few > hours. Including a couple hours to learn how to write c code. How hard can >

Re: [asterisk-users] Force a new user to configure Comedian mail?

Thanks for the tip, all! I forgot that the sample .confs are as much a source of documentation as voip-info.org --JW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Friday, September 14, 2007 2:08 PM To: Asterisk Users Mailing Lis

[asterisk-users] 3 way Calling

Hello, I have recently installed the TrixBOX CE 2.2.4. How can I make calls and use the 3 way calling? can it be done with any IP phone or softphone? should I do any special configuration on TrixBox? Regards, Seysan ___ Sign up now for AstriCon 2007

Re: [asterisk-users] Force a new user to configure Comedian mail?

Jeremy Wadhams wrote: > In Asterisk 1.4, is there any way to "force" new users to configure > their mailbox? I'm thinking a simple IVR that holds a user's hand > through changing their PIN, recording their name, and setting up one > or both greetings, the very first time they use the account. Y

Re: [asterisk-users] Force a new user to configure Comedian mail?

Jeremy Wadhams wrote: > In Asterisk 1.4, is there any way to "force" new users to configure > their mailbox? I'm thinking a simple IVR that holds a user's hand > through changing their PIN, recording their name, and setting up one or > both greetings, the very first time they use the account. >

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

Jared Smith wrote: > On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: >> . matches any number of the preceding character, change it to _X.*X. > > That still won't help. Once the Asterisk pattern matching parser sees a > period in the pattern, it ignores anything after it. (I'm not exact

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On 9/14/07, Jared Smith <[EMAIL PROTECTED]> wrote: > On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: > > . matches any number of the preceding character, change it to _X.*X. > > That still won't help. Once the Asterisk pattern matching parser sees a > period in the pattern, it ignores an

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

Anthony Messina wrote: > I am working on getting freenum.org ISN/ITAD numbers integrated into my > exiting dialplan however I am having trouble getting the extension matches to > work "as expected." > > I would like to be able to do something like: > exten => _X.*.,1,Macro(isn-outbound...) > >

Re: [asterisk-users] Force a new user to configure Comedian mail?

Jeremy Wadhams wrote: > In Asterisk 1.4, is there any way to "force" new users to configure > their mailbox? I'm thinking a simple IVR that holds a user's hand > through changing their PIN, recording their name, and setting up one or > both greetings, the very first time they use the account. Yep

Re: [asterisk-users] Force a new user to configure Comedian mail?

Jeremy Wadhams wrote: > I know I can publish docs that tell them how to use the "0" menu and do > this by hand... but users are lazy and resent documentation. As are Asterisk administrators (sometimes) :-) See the 'forcename' config option in voicemail.conf.sample. -- Kevin P. Fleming Director

Re: [asterisk-users] Force a new user to configure Comedian mail?

On 9/14/07, Jeremy Wadhams <[EMAIL PROTECTED]> wrote: > > In Asterisk 1.4, is there any way to "force" new users to configure their > mailbox? I'm thinking a simple IVR that holds a user's hand through > changing their PIN, recording their name, and setting up one or both > greetings, the very fi

Re: [asterisk-users] Force a new user to configure Comedian mail?

Jeremy Wadhams wrote: > In Asterisk 1.4, is there any way to "force" new users to configure > their mailbox? I'm thinking a simple IVR that holds a user's hand > through changing their PIN, recording their name, and setting up one or > both greetings, the very first time they use the account. >

Re: [asterisk-users] Prompt for extension with standard dial-tone.

On 9/14/07, Jared Smith <[EMAIL PROTECTED]> wrote: > On Fri, 2007-09-14 at 19:49 +0300, Atis wrote: > > What i want to do - is to give ability for answered call to hear > > regular dial tone and be able to enter phone number - that i would > > dial later. > > Does the DISA() application do what you

[asterisk-users] AGI script fails on IAX channels (from call file).

Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and

Re: [asterisk-users] Paging to external speaker like in airports etc...

At 01:11 PM 9/14/2007, you wrote: >Unfortunately, that seems to be more and more the way of the world, >though I will say that this kind of unproductive attitude and >intolerance of others is more prevalent on these kinds of computer >related lists than some other lists. >Something about being able

Re: [asterisk-users] ztdummy kills audio

Yes, that was after approx a minute. Output from lsmod is... zaptel182948 4 zttranscode,ztdummy crc_ccitt 3072 1 zaptel > On Fri, Sep 14, 2007 at 03:32:00PM -0400, John Albano wrote: > >> Opened pseudo zap interface, measuring accuracy... >> --- Results after 0

[asterisk-users] Force a new user to configure Comedian mail?

In Asterisk 1.4, is there any way to "force" new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. I know I can publish docs that t

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On Friday 14 September 2007 12:37:11 pm Tilghman Lesher wrote: > On Friday 14 September 2007 11:39:40 Anthony Messina wrote: > > I am working on getting freenum.org ISN/ITAD numbers integrated into my > > exiting dialplan however I am having trouble getting the extension > > matches to work "as exp

Re: [asterisk-users] ztdummy kills audio

On Fri, Sep 14, 2007 at 03:32:00PM -0400, John Albano wrote: > Opened pseudo zap interface, measuring accuracy... > --- Results after 0 passes --- > Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Is this after a minute? What is the output of: lsmod | grep zaptel --

[asterisk-users] g729 on 1.4.10.1

I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow

Re: [asterisk-users] Paging to external speaker like in airports etc...

Anthony Kepler wrote: > Thats AMAZING! This "google" you have shown me is truly a modern marvel > of the interwebs. > > You know what would be EVEN BETTER though? > If idiots (such as you and I) would find something better to do with our > time than mock others on mailing lists in a pitiful att

Re: [asterisk-users] Paging to external speaker like in airports etc...

Thats AMAZING! This "google" you have shown me is truly a modern marvel of the interwebs. You know what would be EVEN BETTER though? If idiots (such as you and I) would find something better to do with our time than mock others on mailing lists in a pitiful attempt to appear more knowledgeable/

Re: [asterisk-users] ztdummy kills audio

Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 > On Fri, Sep 14, 2007 at 03:05:49PM -0400, John Albano wrote: > >> I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the >> dialplan hang

Re: [asterisk-users] ztdummy kills audio

On Fri, Sep 14, 2007 at 03:05:49PM -0400, John Albano wrote: > I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the > dialplan hangs when it tries to play audio (i.e. Playback) -- and I just > hear static on the line. I'm running this on a debian system. I actually > have it work

[asterisk-users] ztdummy kills audio

I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the dialplan hangs when it tries to play audio (i.e. Playback) -- and I just hear static on the line. I'm running this on a debian system. I actually have it working on a different debian system but have yet to discover the import

Re: [asterisk-users] how to route outgoing calls on IP-level

Check the "route" command on your Linux system. The gateway route should be the ethX and network whatever you want. At 01:41 p.m. 14/09/2007, Drew Gibson wrote: Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to

Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

On Thu, Sep 13, 2007 at 11:55:59PM -0700, Vieri wrote: > Thank you, > I did what you mentioned below. > It seems that I'm getting a hangupcause of 0 which I > believe is "not defined". > Is Alcatel the first party that is trying to > disconnect or is it Asterisk? (Not sure how to > interpret the de

Re: [asterisk-users] Skype + Asterisk

Alejandro, Thanks for replying. I did come by this website before. I was just wandering, if anybody actually tried Skype with Asterisk. My experimentation with the Sip Protocol and Asterisk is at end because I could never get QOS with any sip provider, ie Broadvoice, Vitelity, and Teliax

Re: [asterisk-users] how to route outgoing calls on IP-level

Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate This is the job of your network, no

Re: [asterisk-users] Prompt for extension with standard dial-tone.

On Fri, 2007-09-14 at 19:49 +0300, Atis wrote: > What i want to do - is to give ability for answered call to hear > regular dial tone and be able to enter phone number - that i would > dial later. Does the DISA() application do what you want? -- Jared Smith Community Relations Manager Digium,

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: > . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but tha

Re: [asterisk-users] MOH Files Volume

Peder @ NetworkOblivion wrote: > Is there a way to decrease the volume on the native files version of MOH > in 1.4? I've had several people complain that it is too loud. run the files through sox -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___

Re: [asterisk-users] how to route outgoing calls on IP-level

On Fri, 14 Sep 2007, Kate Kretz wrote: > Dear Sirs, > > out asterisk server has multiple network cards. > > I want some outgoing calls (from several extensions) to use one IP address, > and others to go through > another address. > > is there a way to achive that using asterisk ? I doubt it, but

Re: [asterisk-users] outgoing call restriction in extention.conf

The best way of restricting users from STD is making different context in extensions.conf, in that context allow STD. and in sip.conf for those users make that context. extensions.conf [local] exten => _0[1-9].,1,Answer exten => _0[1-9].,2,Dial(${TRUNK}/${EXTEN:1}w) exten => _0.,3,Hangup [ST

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

i just met the same problem. i want to match extension that end with a number, but can not find a way. i also found that "_.X" match all extension, but won't match any caller-id number in dialplan. maybe it is a bug. but it seems not important since "_.X" is useless anyway. 2007/9/15, Tilghman Le

Re: [asterisk-users] Skype + Asterisk

Did you got a response for your questions? Recently found this URL in Google SiSky http://www.yeastar.com/ProductsforAsterisk.asp Regards, Alejandro Lengua On 9/6/07, John Meksavan <[EMAIL PROTECTED]> wrote: > > Has anybody ever integrated Skype with Asterisk? If you have, which > software would

[asterisk-users] MOH Files Volume

Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. Peder ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocatio

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

On Friday 14 September 2007 11:39:40 Anthony Messina wrote: > I am working on getting freenum.org ISN/ITAD numbers integrated into my > exiting dialplan however I am having trouble getting the extension matches > to work "as expected." > > I would like to be able to do something like: > exten => _X

[asterisk-users] how to route outgoing calls on IP-level

Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate ___ Sign up n

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

I don't think * means anything special to A*k, But wouldn't it be: _X.*X. To match as you ask ? (number)(wildcard)*(number)(wildcard) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: 14 September 2007 17:40 To: Asterisk Users Mail

[asterisk-users] AstLinux 0.4.8 Released

Hello everyone, AstLinux 0.4.8 has been released. The only updates were to Asterisk and Zaptel. Most of the development effort is focused on implementing Asterisk 1.4 and releasing AstLinux 0.5, which should be both happen fairly soon. Expect many more changes in those releases! http://www

Re: [asterisk-users] Can Asterisk match a literal "*" in extensions.conf

. matches any number of the preceding character, change it to _X.*X. Anthony Messina wrote: > I am working on getting freenum.org ISN/ITAD numbers integrated into my > exiting dialplan however I am having trouble getting the extension matches to > work "as expected." > > I would like to be able

[asterisk-users] Prompt for extension with standard dial-tone.

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how

[asterisk-users] Can Asterisk match a literal "*" in extensions.conf

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

Matthew Fredrickson wrote: > shadowym wrote: > >>Maybe his comments were taken out of context as they don't have the whole >>interview posted. Why is he talking about queue games, Biologicall and >>other extremely niche crap when there are huge holes in the basic offering >>(SLA and SCA)? > >

Re: [asterisk-users] Help Drop Calls

Yeah you can do nothing about the routing time out on the PSTN, and there is always a bit of processing time when a call enters the queue. paul aldee wrote: > hi. i have a prob hope someone has a solution for it. here is the > set up local_number ---> another_local_number > --callforward--

[asterisk-users] Help Drop Calls

hi. i have a prob hope someone has a solution for it. here is the set up local_number ---> another_local_number --callforward--> toll free number --> enters asterisk -->queue(agents) the problem is we have lots of drop calls the reason being the original caller puts down his phone before an

Re: [asterisk-users] Agent Callback Login in 1.4

James FitzGibbon wrote: > On 9/13/07, *Kevin P. Fleming* <[EMAIL PROTECTED] > > wrote: > > It shouldn't be that hard to translate the AEL example into > traditional > dialplan language; in fact, Asterisk does that itself when you > load the > AEL int

Re: [asterisk-users] DECT SIP phones

On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote: > Hi folks: > > I know it's come up a few times before, but I need some more detail. > > I'm looking for a SIP DECT (cordless) phone for North American > installations. I've heard only of the Siemens Gigaset S450/C450 phones. > Apparently th

Re: [asterisk-users] Agent Callback Login in 1.4

On 9/13/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > > It shouldn't be that hard to translate the AEL example into traditional > dialplan language; in fact, Asterisk does that itself when you load the > AEL into memory, so if you load it yourself and then do a 'dialplan > show' you'll see the

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

On 9/14/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > Anthony Francis wrote: > > When a device is called and it is in CFWD mode it sends back a redirect > > message (Moved Temporarily), Asterisk displays in the CLI " Recieved > > "Moved Temporarily" trying XX thanks to XXX.XXX.XXX.XXX

Re: [asterisk-users] Asterisk voice quality tuning

I will try to answer it this way: G.711 is toll quality voice, if everything is functioning properly should be almost identical to a regular phone call. You will need to do trouble shooting to (in the words drilled into me by an old boss): isolate, identify and quantify the issue. I would start

Re: [asterisk-users] Mutipoint Conferencing?

On 14 Sep 2007, at 13:45, William Stillwell (Ki4swy) wrote: > I am trying to determine what would need to be done/modified to > enable the following: > > I have a SIP extension come into my asterisk box, and I then need > it to call "6-10" remote Sip Stations that are set to Auto-Answer... >

Re: [asterisk-users] DECT SIP phones

On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote: > I'm looking for a SIP DECT (cordless) phone for North American > installations. I've heard only of the Siemens Gigaset S450/C450 phones. > Apparently these aren't sold for use in NAm, even though they're > supposed to be legal (in the Un

[asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400

Originally posted at http://forums.digium.com/viewtopic.php?t=18045 Hi! I'm trying to configure a DISA setup (A

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

Anthony Francis wrote: > Eric "ManxPower" Wieling wrote: >> SIP response 486 is "Busy Here" according to RFC 3326. Polycoms at >> least (and I think Cisco phones) do not send back a different message >> depending on if DND is enabled .vs. the line appearance simply being busy. >> >> Personally I

[asterisk-users] AsteriskNOW + legacy PBX integration

Hi, I wonder if this question has been answered before, but im kind of stuck.. I have been trying to setup AsteriskNOW with a Digium TDM844B card with 4FXS/4FXO modules.. trying to connect it with a Panasonic KT616 PABX.. this has 6CO ports and 16 extensions. All the extensions are used up, the onl

Re: [asterisk-users] outgoing call restriction in extention.conf

satish patel wrote: > Dear all > >I have asterisk PBX and 100 endpoint i want to block > STD for specific users or password protect so is it possible users can > set passwd on his/her phone and password automaticaly reflacted on > asterisk in short i want to restrict STD call o

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

Eric "ManxPower" Wieling wrote: > SIP response 486 is "Busy Here" according to RFC 3326. Polycoms at > least (and I think Cisco phones) do not send back a different message > depending on if DND is enabled .vs. the line appearance simply being busy. > > Personally I can't see how the people tha

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

To work around our inability to know in advance whether or not, an extension is forwarded or DNDed, we disabled those features (using hardphone settings) and provided a software replacement (which edit database from which Asterisk check user preferences for every call). This is completely "against

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

--- Steve Langstaff <[EMAIL PROTECTED]> wrote: > The OP was asking whether they could update > Asterisk's DND status > > I think that they *actually* want to do some queue > management based on > the DND button of the (SIP) phone. > > > -Original Message- > > From: [EMAIL PROTECTED] >

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

SIP response 486 is "Busy Here" according to RFC 3326. Polycoms at least (and I think Cisco phones) do not send back a different message depending on if DND is enabled .vs. the line appearance simply being busy. Personally I can't see how the people that designed SIP could justify not being ab

[asterisk-users] outgoing call restriction in extention.conf

Dear all I have asterisk PBX and 100 endpoint i want to block STD for specific users or password protect so is it possible users can set passwd on his/her phone and password automaticaly reflacted on asterisk in short i want to restrict STD call of users of outgoing Regards sa

Re: [asterisk-users] Asterisk voice quality tuning

I have both type of call sip-2-pstn and pstn-2 -sip but quality is not good so how to check asterisk voice quality and codec quality i am useing G.711 alaw and ulaw and it is my LAN network so is there any specific perameter or option to improve quality of voice ??? Adrian Marsh <[EMAIL P

Re: [asterisk-users] CallWithUs Service?

On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote: >   I am thinking about selecting CALLWITHUS as my sip provider.  Has anybody > ever used them?  How was the call quality?  DTMF Tones issues? it was your message that prompted me to take a look at callwithus.com. i currently use diam

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

The OP was asking whether they could update Asterisk's DND status for the extension to mirror a DND button on the (SIP) phone. I suggested that they might act on the response code to an OPTIONS. I think that they *actually* want to do some queue management based on the DND button of the (SIP) phon

[asterisk-users] Mutipoint Conferencing?

I am trying to determine what would need to be done/modified to enable the following: I have a SIP extension come into my asterisk box, and I then need it to call "6-10" remote Sip Stations that are set to Auto-Answer... (note, my remote sip stations are actually cisco h323 devices, I can call

Re: [asterisk-users] CallWithUs Service?

> There has to be some reasonable priced sip provider that would perform > just as well as AT&T. Does it exist? The problem is that there is no QoS control between you and the provider, so a lot of the quality issues you have are probably not related to the specific provider, but just the "ge

Re: [asterisk-users] [Serusers] user meeting (beer drinking in Vienna)

Curses! I just got BACK from Vienna yesterday. I should have stayed another week. :) N. Klaus Darilion wrote: > Hi! > > I proudly announce the first ser/openser/asterisk beer drinking evening > in Vienna. > > When: Thursday (thirsty day) 20. September 2007, 19:00 CEST > Where: Vienna, a bar

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

> > --- Steve Langstaff <[EMAIL PROTECTED]> wrote: > > > I don't know about the 1.4 source, but in 1.2 I > > guess you would have to > > add some more code to > > > > handle_response_peerpoke() > > > > to handle the case where you got a 486 response from > > the peer. > > ok thanks, so that ju

Re: [asterisk-users] DECT SIP phones

Håkan Källberg schrieb: > On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: >> I'm looking for a SIP DECT (cordless) phone for North American >> installations. I've heard only of the Siemens Gigaset S450/C450 phones. >> Apparently these aren't sold for use in NAm, even though they're >

Re: [asterisk-users] TE405P intermittent yellow alarm

Thanks to everyone who helped with this. Don Pobanz wrote: > On Thursday, September 13, 2007 4:58 AM Richard van der Hoff said >> Thanks for your help, but again I'd like to ask: what does a yellow >> alarm actually mean? From the driver source code I can see it is set >> when the FRS0 register

Re: [asterisk-users] Asterisk cli

On 9/13/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Thu, Sep 13, 2007 at 10:36:56AM -0500, Mark Michelson wrote: > > Rizwan Hisham wrote: > > > i connect remotely. I have tried both of these cases but no warnings > > > or mesages still. > > > > It could be that your logger.conf file doesn't k

Re: [asterisk-users] Asterisk voice quality tuning

Satish, Whats your network setup? Do you get bad quality on internal-asterisk calls, or only on external calls? Are you making pure IP calls (sip2sip), or are there E1/T1 cards involved? What codecs are you currently using? What devices are you using? Adrian Marsh ___

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

--- Steve Langstaff <[EMAIL PROTECTED]> wrote: > I don't know about the 1.4 source, but in 1.2 I > guess you would have to > add some more code to > > handle_response_peerpoke() > > to handle the case where you got a 486 response from > the peer. ok thanks, so that just seems to confirm that A

Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff

I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Vieri > Sent: 1

[asterisk-users] [SOLVED] fax machine detection for outgoing call on DIVAcard

I was helped by Armin Schindler from the chan_capi user list. So this is my answer and the solution to the chan_capi list : - >Make sure you have enabled the onboard DSP by using > softdtm

Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> wrote: > Looks like the Alcatel is sending back a busy. Note: I'm using libpri patched with BRIstuff. http://ftp.digium.com/pub/libpri/libpri-1.2.4.tar.gz http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-d.tar.gz

Re: [asterisk-users] bug in 1.2.24

> -- Goto (ext-queues,7141,1) > -- Executing NoOp("Zap/9-1", "do not answer call before entering > queue") in new stack > -- Executing SetCIDName("Zap/9-1", "CN") in new stack > -- Executing Set("Zap/9-1", > "

Re: [asterisk-users] DECT SIP phones

On 08:00, Fri 14 Sep 07, H?kan K?llberg wrote: > On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: > > I'm looking for a SIP DECT (cordless) phone for North American > > installations. I've heard only of the Siemens Gigaset S450/C450 phones. > > Apparently these aren't sold for use in