[asterisk-users] How to barge Inbound calls

Hi All Can anybody help me for dial plan which can barge inbound call groupwise. Because when i am trying to barge inbound calls which is coming on my DID number i can hear 1st 3 digit of my Inbound provider IP address instaed of extension which pick that calls. I tried Chanspy as well as

[asterisk-users] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'

Sorry to post a C compile error on this mailing list but this is Asterisk related. Basically, I was following http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu e_logging to patch logger.c and Makefile in Asterisk 1.4.* in order to write queue_log to mySQL database. When I

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

On Thu, 9 Oct 2008, Steve Totaro wrote: > I don't have answers just a question. > > DAHDI is alpha or beta code, what motivates you to upgrade so badly that you > are frustrating yourself so much? Perhaps the fact that zaptel is not listed anymore on the Digium website? :) _

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

On Thu, 9 Oct 2008, Sean Bright wrote: > On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse <[EMAIL PROTECTED]> wrote: >> The information (or lack of it) on upgrading from zaptel to that >> @&*^QW%&^%!!! dahdi is very frustrating. >> >> I cannot find anything on how to uninstall zaptel, i found an ea

[asterisk-users] Be aware of callcheap.com and Mike Low - It is scam

Hi everybody, Recently I was ripped off by this company named Callcheap Networks Inc, and so did one of the carriers I recommended to them. Now I am perusing legal action against them, a mess in which I never wanted to get into. Based on my bad experience, I wanted to let everybody know if this gu

Re: [asterisk-users] How to enable inbound CLI for X-Lite/Asterisk phone.

Quoting "Syed Nasruddin" <[EMAIL PROTECTED]>: > I am using asterisk 1.4.18. I am using it for inbound only call center. > The SIP phones are X-Lite. Right now when a call is proxied by Asterisk > to X-Lite the agent only sees asterisk written on its CLI screen. I want > the agents to be able to vi

Re: [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to`mysql_error'

> This looks really old and weird. I could suggest using realtime > queue_log backport from 1.6 which i'm currently using. That's good info, Atis. I will definitely give it a go. <>___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)

The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. This is documented in this changelog: http://downloads.digium.com/pub/asteri

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

On Fri, Oct 10, 2008 at 3:22 AM, Remco Barendse <[EMAIL PROTECTED]> wrote: > On Thu, 9 Oct 2008, Steve Totaro wrote: > >> I don't have answers just a question. >> >> DAHDI is alpha or beta code, what motivates you to upgrade so badly that you >> are frustrating yourself so much? > > Perhaps the fac

Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)

Brendan Martens wrote: > The reason for this is that 1.6.0 does not support dahdi. It was a > mistake when it was listed as an included feature. The documentation > for it has been removed in 1.6.0.1. If you need dahdi you need to go > to 1.6.1. That is incorrect. There was one small feature

Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)

I see, Thank you for the clarification. Brendan Martens On Oct 10, 2008, at 9:31 AM, Kevin P. Fleming wrote: > Brendan Martens wrote: >> The reason for this is that 1.6.0 does not support dahdi. It was a >> mistake when it was listed as an included feature. The documentation >> for it has been r

Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)

Inbound calls on DAHDI work fine. At some point, outbound starts working. I just cannot figure out what the trigger is. At first, I thought the trigger was receiving at least one inbound call. But that isn't always true. Once it starts working, it seems to continue until a restart. Everything

[asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining.

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

Anthony Messina wrote: > On Thursday 09 October 2008 09:57:30 pm Steve Totaro wrote: >> Now I have not touched any of that code, but to me, it would have been much >> simpler to change names, then change functionality later. Make DAHDI a >> drop in replacement for Zaptel, in fact, if memory serves

Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4

You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Dave -Original Message- From: [EMAIL PROTECTED] [mailto

[asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)

Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the b

[asterisk-users] How to enable inbound CLI for X-Lite/Asterisk phone.

Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make

Re: [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'

On Fri, Oct 10, 2008 at 10:50 AM, Lee, John (Sydney) <[EMAIL PROTECTED]> wrote: > Sorry to post a C compile error on this mailing list but this is > Asterisk related. > > Basically, I was following > http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu > e_logging > > to patch l

Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

On Thu, 9 Oct 2008, Mike wrote: > On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote: >> Mike, >> >> Can you tell us : >> >> - asterisk version >> - zaptel version >> >> When you call over this line, when you hangup did you hear an busy >> tone ? or any class tone ? To do this test conne

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

Steve Totaro wrote: > > > On Thu, Oct 9, 2008 at 10:32 PM, sean darcy <[EMAIL PROTECTED] > > wrote: > > Remco Barendse wrote: > > The information (or lack of it) on upgrading from zaptel to that > > @&*^QW%&^%!!! dahdi is very frustrating. > > >

Re: [asterisk-users] Howto analyze concurrent ISDN channel usage

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Thursday, October 09, 2008 2:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Howto analyze concurrent ISDN channel usage Hi, Does anyone have a suggestion how I can analyze

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting "ulimit -n 32768", but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Any suggestion?? On Fri, Oc

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

On 10/10/08, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > After getting some ERRORS like this: > > [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup > media stream for this call. > [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup > media stream for this

[asterisk-users] Question about echo cancelation

Hi, I'm using the following setup : Alice IPPhone --- Media gateway --- Phone Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engin

[asterisk-users] softclient for customized apps like a call center?

There are a variety of open source (and closed source) software-based Windows SIP or IAX phones out there. However, I am thinking of using one for a inbound call center. Some of the things I'd be looking for: The ability to make/receive calls (duh!). The ability to 'launch' a web browser ba

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

On 10/10/08, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > Kristian: > > Thanks for your reply. I am running asterisk as root, but still getting this > error. > > I did a test while running asterisk 1.4.21 version setting "ulimit -n > 32768", but after restaring asterisk it stop working with less tha

[asterisk-users] Got event 17 (Polarity Reversal)...

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as

[asterisk-users] Block Caller ID

Hi Is there any way to stop Asterisk from sending Caller ID display on the softphones ? I;ve E1 PRIs and SIP extensions , i need to stop caller ID from appearing on the softphones ...but in CDRs caller Ids should show - so please dont suggest to set "blockcallerid=yes" in zapata.conf ;) Than

Re: [asterisk-users] Transfer/Park Question.

Daniel Hazelbaker wrote: > On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote: > > > Short answer: currently no. > > Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and > we do call parking with DTMF. People were used to just hitting PARK > and their phone displaying the park

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

You should not get that message on analog lines in the USA or Canada. I suspect your line has a provisioning issue or is using different signaling than you think it is using. Jim Duda wrote: > Can anyone tell me what this message means? > > Got event 17 (Polarity Reversal)... > > I'm running

Re: [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam

On Fri, Oct 10, 2008 at 5:09 AM, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: > Hi everybody, > > Recently I was ripped off by this company named Callcheap Networks Inc, and > so did one of the carriers I recommended to them. Now I am perusing legal > action against them, a mess in which I never wan

Re: [asterisk-users] Question about echo cancelation

All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Echo must be removed before the call is converted to VoIP -- in your ca

Re: [asterisk-users] Howto analyze concurrent ISDN channel usage

Luis Morales wrote: > Try with fop, > > http://www.asternic.org/ Thanks Luis. I'll give that a try. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update option

Re: [asterisk-users] Howto analyze concurrent ISDN channel usage

Jay Taylor wrote: > Hi, > > It may not be that you are out of channels. I've recently tried to setup my > ISDN line for use with asterisk and ran into a similar issue. Some people > could call me and others couldn't. My asterisk box was rejecting some calls > with an "Incompatible Destinat

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

On Fri, Oct 10, 2008 at 12:57:27PM -0400, Jim Duda wrote: > Can anyone tell me what this message means? > > Got event 17 (Polarity Reversal)... > > I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. > > It appears that I get this Polarity Reversal each time an inbound call > han

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez wrote: > Kristian: > Thanks for your reply. I am running asterisk as root, but still getting this > error. > > I did a test while running asterisk 1.4.21 version setting "ulimit -n > 32768", but after restaring asterisk it stop working with

Re: [asterisk-users] Howto analyze concurrent ISDN channel usage

Stefan Schmidt wrote: > you could use mrtg to get stats of the overall usage of the server. or Thanks for your suggestion. I found a script here: http://karlsbakk.net/asterisk/ Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Transfer/Park Question.

Brent Davidson wrote: Also be aware that in 1.2.x and 1.4.x, if you park a call and then pick it up, you can't park it again. At least not with the DTMF I wasn't aware of the inability to re-park calls in 1.4 That could have been a nasty surprise. I would be very interested in the

Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4

On 08:26, Fri 10 Oct 08, David Gibbons wrote: > You need to check out the chan_sccp-b mainling lists on sourceforge. There is > active development in SVN but not in tarball releases. > > http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion > > It is very stable. Or, if

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

>You should not get that message on analog lines in the USA or Canada. I >suspect your line has a provisioning issue or is using different >signaling than you think it is using. Not necessarily true. Most recent solid state switches have abandoned this as a cost saving measure. Polarity rev

Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

Having 600 channels it would be like 1200 RTP ports. And on the rtp.conf I have fonfigured from 1 to 2. I do not think this is the problem. Thanks, Juan On Fri, Oct 10, 2008 at 1:38 PM, Tzafrir Cohen <[EMAIL PROTECTED]>wrote: > On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

Hi Jim, We had this exact problem with our system for several years. A call would come in with no caller ID and when we answered nobody would be there. On the Asterisk console would be the "Got event 17 (Polarity Reversal)" message. We spent hours and hours on this. Our carrier was AT&T (SBC).

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

Dan Peters wrote: > interesting thing is that we didn't get the calls all the time but when > we did get them they were ALWAYS on the hour or half-hour. > Sounds like it may have been a line test. I vaguely recall a thread going on here about such tests causing issues. Doug -- Ben Frankl

Re: [asterisk-users] Menu for call forwarding or voicemail

> I would like to create a simple menu that would allow a caller to > decide whether they want to leave a message or be forwarded to another > number (i.e cell phone). Thanks in advance for any insight. > > Here's my current extension.conf > > [general] > static=yes > writeprotect=yes > > [globals]

Re: [asterisk-users] Transfer/Park Question.

Doug Lytle wrote: > Brent Davidson wrote: >>> Also be aware that in 1.2.x and 1.4.x, if you park a call and then >>> pick it up, you can't park it again. At least not with the DTMF >>> >> I wasn't aware of the inability to re-park calls in 1.4 That could >> have been a nasty surprise.

Re: [asterisk-users] Transfer/Park Question.

Brent Davidson wrote: > Ok, the patch is working great. Any idea what would make the one step > parking not work? I've tried several DTMF combinations in features.conf > Check your featuredigittimeout, it defaults to 1/2 second. You may need to increase it. I have mine set to ## to acti

[asterisk-users] Budge Tones pick up wrong calls

We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respon

Re: [asterisk-users] Transfer/Park Question.

On Oct 10, 2008, at 1:00 PM, Brent Davidson wrote: > Doug Lytle wrote: >> I don't remember where I got it (Might have been the bug tracker) >> that >> works fine under the current 1.4.x. I had to do a minor change to >> get >> it to apply. >> >> Copy into Asterisk source directory >> >> patch

Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4

Thanks both, The only thing I have a little concern over is that 1.6 is that its still a development release (if I understand things correctly). Stability is the main thing for me (its only a very small set up) but there are no technical people around if something were to go wrong through the

Re: [asterisk-users] Transfer/Park Question.

Doug Lytle wrote: Brent Davidson wrote: Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf Check your featuredigittimeout, it defaults to 1/2 second. You may need to increase it. I hav

Re: [asterisk-users] Transfer/Park Question.

Daniel Hazelbaker wrote: You won't. The patch I sent you off-list is incomplete, this one is better. I forgot I fixed the "parked has timed out" option in another patch before I fixed this part. Anyway, make sure when you dial you put "k" in the dial options ("K" too if you want both si

[asterisk-users] Help need for debuging the core file.

I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for

[asterisk-users] is there a way

hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhost

[asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that te

Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4

On 21:28, Fri 10 Oct 08, Wayne wrote: > Thanks both, > > The only thing I have a little concern over is that 1.6 is that its > still a development release (if I understand things correctly). No, 1.6.0 has been released. This is indeed the first public 'final' release of the 1.6 series. But it's

Re: [asterisk-users] Help need for debuging the core file.

On Friday 10 October 2008 15:42:34 gary wrote: > I am running asterisk 1.2.27 and it dead today. The following is the > backtrace of core file. Can anybody help me to identify what is the > possible cause of crash? It seems the mysql connection causing problem in > Thread 2. But I can not tell what

Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <[EMAIL PROTECTED]>wrote: > Hello, > > > > We have 2 SIP trunks from Bandwidth.com and if both are in use and someone > tries to dial out, they cause another call to get one-way audio (the caller > hears us, we cannot hear them). This happens 100% of t

Re: [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam

I also thought about it. Maybe I should not have posted it here. But I know he is actively searching for another company. Just don't want any other provider to suffer. Zeeshan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ast

[asterisk-users] Caller ID service and the ethernet stucking

Hi All; We added the callerid service on our telephone line, once that done, now when we call to the Asterisk PBX or we need to place outside call via the digium (zaptel channel), the PBX got a problem in the network, and we become not able to reach it, this stay for a while of time (about 5 mi

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Jim Tzafrir Cohen wrote: > On Fri, Oct 10, 2008 at 12:57:27PM -0400, Jim Duda wrote: >> Can anyone tell me what

Re: [asterisk-users] is there a way

Babcock, Michael Alex wrote: > hey; > i'm at best western and am curious is there a way i could find out if > our best western, with out asking, is using asterisk? > oh and petsmart i think is using asterisk they have alason voice for > there main voicem enu. > mike > > > thanks for reading > S

Re: [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam

On Fri, Oct 10, 2008 at 5:43 PM, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: > I also thought about it. Maybe I should not have posted it here. But I know > he is actively searching for another company. Just don't want any other > provider to suffer. > > Zeeshan > > Again, your motives are admirabl

Re: [asterisk-users] is there a way

On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson <[EMAIL PROTECTED] > wrote: > Babcock, Michael Alex wrote: > > hey; > > i'm at best western and am curious is there a way i could find out if > > our best western, with out asking, is using asterisk? > > oh and petsmart i think is using asterisk they

Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

Hi Steve, It's behind a NAT/Firewall but SIP translation is enabled and removing it from behind the firewall did nothing, it still dropped calls. The calls connect and everything works, but it dies when all trunks are in use and someone else tries to call out. It seems like even though both channe

Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

You need to configure your box for nat settings, externip and other settings in sip.conf and set nat=yes instead of nat=no. One way audio is almost always a NAT issue and those are two glaring things that would cause problems. Thanks, Steve Totaro On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen <[

Re: [asterisk-users] Menu for call forwarding or voicemail

> Any reason not to ring both at once? > exten => 101,1,Dial(SIP/101&SIP/[EMAIL PROTECTED],20) > -Darren That would also work but what if my sip/101 device (softphone) isn't connected. Currently if my softphone is not connected then the line will go straight to voicemail. If I remove the voicemai

Re: [asterisk-users] Budge Tones pick up wrong calls

First of all, are the handsets using distinct SIP peers? Are they set up statically or to register? Secondly, unless you are using an Ethernet hub, SIP signaling data destined for one phone should not go to another. Paul Douglas Franklin wrote: > We have 3 Grandstream Budge Tone 100 phones wh

Re: [asterisk-users] is there a way

no i'm a guest at the bestwestern On Oct 10, 2008, at 1:55 PM, Brent Davidson wrote: > Babcock, Michael Alex wrote: >> hey; >> i'm at best western and am curious is there a way i could find out if >> our best western, with out asking, is using asterisk? >> oh and petsmart i think is using asterisk

Re: [asterisk-users] is there a way

steve; thanks a lot mike On Oct 10, 2008, at 2:20 PM, Steve Totaro wrote: On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson <[EMAIL PROTECTED] > wrote: Babcock, Michael Alex wrote: > hey; > i'm at best western and am curious is there a way i could find out if > our best western, with out as

Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

externip messes up DTMF detection, and by messes up I mean it doesn't detect it at all. Setting nat=yes or nat=no didn't make a difference either. When the trunks are in use, the calls are fine, no dropped audio. It only happens when a 3rd call is made and there's no trunk available. Thanks :) O

Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

Oh, I thought you had logic to count the calls on the trunk. You should limit each trunk to one call. This is the primary reason besides the email that basically said that customer support structure has been changed and anything beyond the Demarc would not be supported, I the Demarc is simply the

[asterisk-users] Mitel 5220 firmware

Hi, Just wondering if anyone here has the latest SIP firmware for the Mitel 5220. I have a few of these phones with the latest firmware (6.0.0.19) that is on the Mitel site. However I believe the latest version is 7.2 and they only make it available to partners and resellers. Anyone out

Re: [asterisk-users] is there a way

nmap for scanning and identification. cross platform and even a nice gui for windows. Eric On Fri, Oct 10, 2008 at 3:20 PM, Steve Totaro < [EMAIL PROTECTED]> wrote: > > > On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson < > [EMAIL PROTECTED]> wrote: > >> Babcock, Michael Alex wrote: >> > hey; >>

Re: [asterisk-users] Budge Tones pick up wrong calls

Are you using NAT? On Fri, Oct 10, 2008 at 4:24 PM, Paul Douglas Franklin <[EMAIL PROTECTED]> wrote: > We have 3 Grandstream Budge Tone 100 phones which are being very fluid > on incoming calls. They are set up as extensions 2501, 2518, and 2536. > When calling out to another phone, they always

Re: [asterisk-users] is there a way

I will look into that when I get my Acer Aspire One running FC8, it came with windows XP and I got the 1gig ram, 120gig HD. I am following threads on howto but nobody has a definitive guide yet, that allows the embedded webcam and the NIC to work properly. Maybe (probably) my USB Alpha AWUS036H w

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: > Tzafrir, > > Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect > this will solve my issue. I never would have know to look for this. > > Thanks much! You made my day :-) Hmm... I might have misled you. By default

Re: [asterisk-users] is there a way

On Fri, Oct 10, 2008 at 07:33:45PM -0700, Eric Fort wrote: > nmap for scanning and identification. cross platform and even a nice gui > for windows. What nmap does is called "fingerprinting". it mostly uses the fact that when faced with normal behaviours, most stacks behave the same. But when fac