Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)

On Wed, Oct 01, 2008 at 07:03:06PM -0400, Steve Totaro wrote: > I own this combination of 1s and 0s. 111010010010101001001. Now please, 0x1B1254F81F , what's so novel about this? What is it supposed to do? Why is it not trivial? If you attempts to mock patents you won't get very f

Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)

You need to lighten up buddy. The point is, who owns a series of ones and zeros??? Is your series of one's and zeros better than someone else' ones and zeros? Why because you have more, or the order is different? Take a vacation, get some tail or something. Geez. You are always riding posts,

[asterisk-users] B410p question

Hello list, I have got an asterisk box installed working ok with an b410p card to make and receive isdn calls. All works ok, but when a call is answer and the person starts to speak, always I can ear a "beep" during the call. This beep is ear some times in about 30 seconds between each beep. Past

Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)

On Thu, Oct 02, 2008 at 04:23:26AM -0400, Steve Totaro wrote: > You need to lighten up buddy. The point is, who owns a series of ones and > zeros??? Is your series of one's and zeros better than someone else' ones > and zeros? Why because you have more, or the order is different? You, for insta

[asterisk-users] How to find the CDR call start time value

Can anyone suggest how I can find the value of the call start time that will be logged by CDR in the dialplan? I've taken a look through the variables but I can't see anything that seems to hold this? The information contained in this email is intended for the personal and confidential u

[asterisk-users] DTMF Problem

Dear Sir, I have the following Scenario: 1- I have a DID number from Voxbone mapped to my asterisk server with RFC 2833 protocol used for DTMF 2- On asterisk Server I configured an incoming peer that receives calls from VoxBone and send calls to a2billing context as follow: *sip.conf* [sip_proxy

Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)

On Wed, Oct 1, 2008 at 5:37 PM, tic tac <[EMAIL PROTECTED]> wrote: > Thanks, in my case though it looks like the originating party (polycom > softphone) is hearing a clipped voicemail prompt because of that; should I > look into having that fixed into their firmware? As a workaround, I was > thinki

Re: [asterisk-users] How to find the CDR call start time value

HI Steven, You can get call start time by ${CDR(start)} . For more information of asterisk variables , please check out http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List Thanks, Krunal Patel On Thu, Oct 2, 2008 at 3:08 PM, Steve Hanselman <[EMAIL PROTECTED]>wrote: > Can anyone

Re: [asterisk-users] ATA for large networks

I personnlay found that marc is better than google when searching mailing lists : http://marc.info/?l=asterisk-users&r=1&w=2 > What is the best-recommended resource for searching archives of this > mailing > list? > > Thanks for your time ___ -- Ban

Re: [asterisk-users] How to find the CDR call start time value

That's exactly what I was looking for, I'd found this http://www.voip-info.org/wiki/view/Asterisk+variables which seems to be a partial copy of the same thing. Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Krunal Patel Sent: 02 Octob

Re: [asterisk-users] rebooting snoms in 1.6

> With Asterisk 1.4 I could use commands like: > > /usr/sbin/asterisk -rx "sip notify reboot-snom mjc_home" > > to reboot a snom phone. Now, with 1.6, when I try that, I get: > > Unable to find notify type 'reboot-snom' > Command 'sip notify reboot-snom mjc_home' failed. > > Do I need to add so

[asterisk-users] Ultramonkey LVS + asterisk

hi, has anyone implemented ultramonkey with asterisk? do i really need to setup fwmark as discussed in the url below? thanks! http://www.gossamer-threads.com/lists/lvs/users/20871 regards, ron ___ -- Bandwidth and Colocation Provided by http://www.a

[asterisk-users] IP address on mysql cdr

hi, is it possible to store the IP address of the caller in the CDR? how about the end date/time? thank you. regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Ari

Re: [asterisk-users] rebooting snoms in 1.6

"Dr. Michael J. Chudobiak" <[EMAIL PROTECTED]> writes: > With Asterisk 1.4 I could use commands like: > > /usr/sbin/asterisk -rx "sip notify reboot-snom mjc_home" > > to reboot a snom phone. Now, with 1.6, when I try that, I get: > > Unable to find notify type 'reboot-snom' > Command 'sip notify r

[asterisk-users] DTMF issue

Dear All, What could be the problem if I try to send DTMF in RFC2833 format to my asterisk server and it replies back with 603 error message? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22

Re: [asterisk-users] Cisco Dropping SIP support?

Michael Graves schrieb: > Earlier today I glanced at Junction Networks blog and was surprised to > find a post indicating that Cisco was dropping SIP support in their > 79xx series phones. Here's t > link: > > http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo > rks-lab-cisco

Re: [asterisk-users] How can Block a pri channel

Dwayne Hubbard wrote: > Sean is correct I *never* get tired of hearing/reading that. -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Registe

Re: [asterisk-users] How can Block a pri channel

On Thu, Oct 2, 2008 at 9:40 AM, Sean Bright <[EMAIL PROTECTED]> wrote: > Dwayne Hubbard wrote: > > Sean is correct > > I *never* get tired of hearing/reading that. > > -- > Sean Bright > [EMAIL PROTECTED] > > Insurance is a huge scam. They are betting for you by taking your money, and you are bet

Re: [asterisk-users] Asterisk - Failover System

Redfone is not much good unless you have more than one Asterisk box. Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Oct 1, 2008 at 10:47 PM, Darren Sessions <[EMAIL PROTECTED]>wrote: > I agree that an OpenSER solution on top of Asterisk for a 120 u

[asterisk-users] Zaptel-1.4.1 error cross compile

I have CLFS ARM cross toolchain with uClibc and I have installed asterisk on it now I want to compile zaptel-1.4 I got this error clfs:/mnt/clfs/sources/zaptel-1.4.1$ make make[1]: Entering directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect' checking build system type... i686-pc-linux-gnu ch

Re: [asterisk-users] Zaptel-1.4.1 error cross compile

On Thu, Oct 02, 2008 at 10:19:00AM -0400, satish patel wrote: > > > I have CLFS ARM cross toolchain with uClibc and I have installed asterisk on > it now I want to compile zaptel-1.4 > > I got this error > > clfs:/mnt/clfs/sources/zaptel-1.4.1$ make > make[1]: Entering directory `/mnt/clfs/sou

Re: [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)

Hello, Has anyone successfully used this SIPPEER function ? exten => _753X,n,Set(foo=${SIPPEER(${EXTEN}:curcalls)}) Then, did you get a meaningful value ? I suspect my understanding of it is incorrect as I would say that if an extension is on call with someone else, curcalls shall return 1 (which

Re: [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)

Olivier wrote: > Hello, > > Has anyone successfully used this SIPPEER function ? Olivier, I wasn't able to do any testing on it last night, I'll give it a try over this upcoming weekend and let you know what I find. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to

Re: [asterisk-users] Zaptel-1.4.1 error cross compile

Regards, Satish Patel Quoting Tzafrir Cohen <[EMAIL PROTECTED]>: > On Thu, Oct 02, 2008 at 10:19:00AM -0400, satish patel wrote: >> >> >> I have CLFS ARM cross toolchain with uClibc and I have installed asterisk on >> it now I want to compile zaptel-1.4 >> >> I got this error >> >> clfs:/mnt/c

Re: [asterisk-users] DTMF Problem

On Oct 2, 2008, at 5:27 AM, michel freiha wrote: Dear Sir, I have the following Scenario: 1- I have a DID number from Voxbone mapped to my asterisk server with RFC 2833 protocol used for DTMF 2- On asterisk Server I configured an incoming peer that receives calls from VoxBone and send call

Re: [asterisk-users] Zaptel-1.4.1 error cross compile

On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote: > > Quoting Tzafrir Cohen <[EMAIL PROTECTED]>: As I wrote: > >Could you please try a newer version of zaptel 1.4? There have been many > >changes in the build system of zaptel 1.4 since 1.4.1 . But in your reply: > clfs:/mnt/clfs/so

Re: [asterisk-users] Asterisk custom functions

On Wednesday 01 October 2008 23:58:41 Max Alex wrote: > Hi All, > i have centos5 system, i have installed asterisk 1.4 branch. > i havedone realtime connection with odbc to pgsql. > i have created custom functions in func_odbc.conf, all dsn setup and > connection is working fine, > but custom funct

Re: [asterisk-users] Zaptel-1.4.1 error cross compile

Regards, Satish Patel Quoting Tzafrir Cohen <[EMAIL PROTECTED]>: > On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote: >> >> Quoting Tzafrir Cohen <[EMAIL PROTECTED]>: > > As I wrote: > >> >Could you please try a newer version of zaptel 1.4? There have been many >> >changes in the bu

Re: [asterisk-users] zap destroy

- "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote: > On Wed, Oct 01, 2008 at 01:39:29PM -0500, Jeff Peeler wrote: > > > Nope, that's the best you can do without restarting Asterisk. Is > > requiring two restarts reproducible? I'd really like to see console > > > output with verbosity and debug s

Re: [asterisk-users] zap destroy

On Oct 2, 2008, at 9:10 AM, Jeff Peeler wrote: > > - "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote: > >> Yes, the new changes will be in 1.4.22. I continually have to >> remind myself that users aren't running the most up to date code. Once 1.4.22 comes out I will report if I am still having th

[asterisk-users] OT - Is sip.instance useful ?

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers ___

[asterisk-users] Asterisk Queue question

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest C

Re: [asterisk-users] OT - Is sip.instance useful ?

Olivier wrote: > Hi, > > I've seen some hardphones or Softswitchs now support this sip.instance > feature : > http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt > > I don't really see any convincing use of this draft but I would be > curious to share thoughts on it. > >

Re: [asterisk-users] Asterisk Queue question

On Thu, Oct 2, 2008 at 7:32 PM, voip crazy <[EMAIL PROTECTED]> wrote: > When the asterisk a queue reset their counters? > > I 'm talking about this kind of info in asterisk console. > >>show queue 600 > 600 has 0 calls (max unlimited) in 'ringall' strategy (4s > holdtime), W:0, C:14, A:8,

Re: [asterisk-users] OT - Is sip.instance useful ?

2008/10/2 SIP <[EMAIL PROTECTED]> > Olivier wrote: > > Hi, > > > > I've seen some hardphones or Softswitchs now support this sip.instance > > feature : > > > http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt > > > > I don't really see any convincing use of this draft but

Re: [asterisk-users] DTMF Problem

Hi; This problem I suffered from it for long time, it needs some work from ur side to resolve it, I will give u all the factors that will help u to fix it, and u need to work on it one after one in care: 1) Disable x-windows, gnome, ... at least for all testing. This is very important to be do

Re: [asterisk-users] Zaptel-1.4.1 error cross compile

Regards, Satish Patel Quoting Tzafrir Cohen <[EMAIL PROTECTED]>: > On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote: >> >> Regards, >> >> Satish Patel >> >> >> Quoting Tzafrir Cohen <[EMAIL PROTECTED]>: >> >> > On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote: >> >> >>

[asterisk-users] dahdi-linux 2.0.0 and dahdi-tools 2.0.0 released

The Asterisk development team is pleased to announce the first offical release of the Digium Asterisk Hardware Device Interface (DAHDI). The list of packages released today includes: dahdi-linux 2.0.0 dahdi-tools 2.0.0 dahdi-linux-complete 2.0.0+2.0.0 Both dahdi-linux and dahdi-tools are required

[asterisk-users] Asterisk 1.4.22 and 1.6.0 Released

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. = === Asterisk 1.4.22 = = A

[asterisk-users] Channels crossing...

I have a customer that is reporting that sometimes when they dial an outside line they can hear other conversations. At this moment I am assuming it only happens when they dial an outside number and not between extensions. They are using Asterisk 1.4.11, Zaptel 1.2.12.1 (just upda

Re: [asterisk-users] Channels crossing...

Carlos Chavez wrote: > Is this a problem with Asterisk or Zaptel? How can I fix it? > Actually, sounds like an analog line with a bad punch down or frayed shielding and your getting cross talk. It may also be an issue at your provider. Doug -- Ben Franklin quote: "Those who wo

[asterisk-users] VOIP Provider

Hi All, Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We need flat rate billing per line/trunk, trunking, did’s, and iax or G.729 compatibility. Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.7.5/

Re: [asterisk-users] VOIP Provider

2008/10/2 Gregory Malsack <[EMAIL PROTECTED]> > Hi All, > > > > Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We > need flat rate billing per line/trunk, trunking, did's, and iax or G.729 > compatibility. > > > > Thanks, > > Greg > > No virus found in this outgoing mess

[asterisk-users] DTMF

How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk Queue question

Yes it is, every counter is set to zero: asterisk -rx "module reload app_queue.so" Regards, Daniel Arohuanca t.+51 1 994149553 Peru On Thu, Oct 2, 2008 at 12:05 PM, Atis Lezdins <[EMAIL PROTECTED]> wrote: > On Thu, Oct 2, 2008 at 7:32 PM, voip crazy <[EMAIL PROTECTED]> wrote: > > When the aster

Re: [asterisk-users] VOIP Provider

Hi. We recommend Fonet Global, they work with Asterisk many years ago and provide sip termination, DIDs, etc. At 03:39 p.m. 02/10/2008, Steve Totaro wrote: 2008/10/2 Gregory Malsack <[EMAIL PROTECTED]> Hi All, Can anyone recommend a good VOIP provider in the

Re: [asterisk-users] Channels crossing...

I have seen and heard the recordings to prove crossed calls between people and agents in a busy call center, it was kind of funny with two agents trying to figure out what was going on and a very confused customer. I certainly would not rule out Asterisk, but you always start with the cables. On

[asterisk-users] dahdi service start

I just downloaded the dahdi release and installed it. I removed anything zaptel I could find. /sbin , /lib/modules and others... when doing a service dahdi start it is still looking to ztcfg Why? I have look all about and cant determine why? service dahdi start Loading DAHDI hardware module

Re: [asterisk-users] Cisco Dropping SIP support?

They are probably referring to the fact that the base 7960 is End of Life and the 7960G is probably going to be EOL soon as well, so they won't offer new firmware at the EOL milestone. They have been replaced by the 7961. Completely different firmware and configuration, but there still is sup

[asterisk-users] uninstalling zaptel

What is the correct way to uninstall zaptel in the zaptel directory I can do "make uninstall-modules" which does just that but what about all the other files??? /etc/udev/rules/XX /etc/init.d/XX /sbin/ztXX and others doing a "make uninstall" gives an error. Is there anything that removes all t

Re: [asterisk-users] Channels crossing...

Hey Steve, it's been a while... On Thu, Oct 2, 2008 at 8:33 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > I have seen and heard the recordings to prove crossed calls between people > and agents in a busy call center, it was kind of funny with two agents > trying to figure out what was going on and

Re: [asterisk-users] dahdi service start

Jerry Geis wrote: > wct4xxp: sh: /sbin/ztcfg: No such file or directory > FATAL: Error running install command for wct4xxp >[FAILED] Hmm.. Something in /etc/modprobe.conf, /etc/modules.conf, or /etc/modprobe.d/? _

[asterisk-users] t1 cards

I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can

Re: [asterisk-users] t1 cards

On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use P