Re: [asterisk-users] SIP client floods port 5060 and gets blocked

On Thu, Oct 28, 2010 at 1:42 AM, Jonas Kellens wrote: > On 10/28/2010 12:52 PM, Gordon Henderson wrote: > > On Thu, 28 Oct 2010, Jonas Kellens wrote > >> On 10/28/2010 10:44 AM, Kevin Keane wrote: > >> > >>> I assume that you checked and the remote IP is a legitimate IP phone? > If > >>> not, it c

Re: [asterisk-users] res_fax

On Jan 20, 2011, at 8:53 PM, Steve Underwood > On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: >> On 01/20/2011 11:47 AM, Steve Underwood >> On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: >> > On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: >> >> On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: >>

Re: [asterisk-users] res_fax

On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: > On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: >> On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: >>> I am working on some fax tools for some of my users. I a

Re: [asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection

On 01/20/2011 03:29 PM, David Cunningham wrote: Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. As the description of this mailing list says, it is for *NON-

Re: [asterisk-users] Asterisk to asterisk t.38

Amit Make sure that the trunk you have between the two servers has the t.38 enabled on it. Do you have any NAT between the two servers or are they on the same lan. We do the t.38 faxing between 1.4 and 1.6 asterisk boxes all of the time. Our audio codes gateway dumps into a 1.4 box and all faxe

Re: [asterisk-users] Asterisk to asterisk t.38

On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote: > On 1/20/2011 3:07 PM, Tom Rymes wrote: >> On 01/20/2011 4:26 PM, Amit Nepal wrote: >> >>> I have an Audio code gateway between two asterisk servers. The audio >>> code has PRI connected for PSTN. I can send faxes and receive faxes in >>> ast 1.4 . A

Re: [asterisk-users] Asterisk to asterisk t.38

Yes Tom, I am sending via the PSTN gateway which is audio code in my case. Thank You Amit Nepal On 1/20/2011 3:07 PM, Tom Rymes wrote: On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can se

Re: [asterisk-users] res_fax

On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: > On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: >> On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: >>> I am working on some fax tools for some of my users. I am reading the >>> https://wiki.asterisk.org docs f

Re: [asterisk-users] Asterisk to asterisk t.38

On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is se

Re: [asterisk-users] Ghost ringing

Sorry, this got buried in my inbox. Did you find a fix or the firmware? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 6:05 PM To: Asterisk Users Maili

[asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection

Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specific

Re: [asterisk-users] Asterisk to asterisk t.38

Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4

Re: [asterisk-users] sip dos question

Hi Kyle, On 01-20-2011 20:41, Kyle Kienapfel wrote: I understood that option worked the other way around so attacker thinks peer name is invalid even when they hit a real one. sorry, it must be because i'm not a native english speaker but i don't exactly get what you mean by the above. to

[asterisk-users] Asterisk 1.8.2.2 Now Available (Security Release)

The Asterisk Development Team has announced a release for the security issue described in AST-2011-001. Due to a failed merge, Asterisk 1.8.2.1 which should have included the security fix did not. Asterisk 1.8.2.2 contains the the changes which should have been included in Asterisk 1.8.2.1. This

Re: [asterisk-users] Polycom 500 / MWI

> I've got the following in my phone.cfg: > > > msg.mwi.1.subscribe="" > > > The actual config looks good, but the structure of the XML is off. Here's what I use (and it works): You seem to be missing the closing statement, so your XML is not well formed. Also, I don't know what

Re: [asterisk-users] Asterisk to asterisk t.38

On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal wrote: > I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can > send recieve faxes from both boxes fine to and from pstn. But the faxing > between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between

[asterisk-users] Asterisk to asterisk t.38

I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal -- __

Re: [asterisk-users] sip dos question

I understood that option worked the other way around so attacker thinks peer name is invalid even when they hit a real one. On Wed, Jan 19, 2011 at 2:23 AM, wrote: > Hi List, > > i've been receiving several sip registration probes in the last month, and > as this server is a testing site (no ext

Re: [asterisk-users] context problem

> I may be wrong here, but I think you can only register once. The last > registration received will overwrite the first one. You will need to > specify a second entry and register that one separately. This is the > same reason you cannot register two devices to the same extension. Yes, that's

Re: [asterisk-users] Polycom 500 / MWI

On Thu, Jan 20, 2011 at 12:55 PM, Brian C. Huffman wrote: > Does anyone know how to setup this phone to work with asterisk so that the > indicator light comes on when there's a new message and goes off quickly > (less than a minute) after the message is deleted? > > Thanks, > Brian Brian, I'm us

Re: [asterisk-users] Mailing list question

On 01/20/2011 11:16 AM, Andrew Thomas wrote: Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) API provides the physical services and bandwidth for the mailing lists, but does not operate them. If you go to the lists.digium.com site and

Re: [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny

On Thursday 20 Jan 2011, JR Richardson wrote: > Hi All, > > I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the > asterisk daemon not the safe_asterisk daemon so when asterisk is > running and I ssh tot he server then 'asterisk -vr' to attach to the > asterisk console there are no

Re: [asterisk-users] Polycom 500 / MWI

Brian C. Huffman wrote: Does anyone know how to setup this phone to work with asterisk so that the indicator light comes on when there's a new message and goes off quickly (less than a minute) after the message is deleted? My phone.cfg for extension 4221 and the voicemail extension of 4200 lo

Re: [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny

On 20 January 2011 18:01, JR Richardson wrote: > Or is there another work around to get ssh console colors using the > Debian * 1.6.0.28 LSB init script? > > I also tried 'nocolor = no' in the [options] section of asterisk.conf > with no effect. > Try running asterisk using safe_asterisk.. Wo

[asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny

Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start aste

[asterisk-users] Polycom 500 / MWI

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: msg.mwi.1.subscribe="" > and the indicator will come o

[asterisk-users] iNum at 12 Noon EST Friday

Hi, Tomorrow, our discussion is around iNum with lots of interesting people chiming in, including the Voxbone people who manage the space. If you ever wondered about iNum and why you might care about it, how it works, who offers it and who actually uses it, here's a chance to find out more. Join

Re: [asterisk-users] Internode weirdness

On 01/19/2011 10:34 PM, Da Rock wrote: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up. Have you tried disallowing re-invites? -- ___

Re: [asterisk-users] Mailing list question 2

Tell you what Steve - I'll not take you up on your kind offer - I'll just let my server keep adding the disclaimer. There - problem solved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 20

Re: [asterisk-users] Mailing list question

Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 20 January 2011 17:04 To: 'Asteri

Re: [asterisk-users] Mailing list question

On Thu, Jan 20, 2011 at 12:03 PM, Danny Nicholas wrote: > Putting the "--" in front of it might make it go away. If I am not mistaken it should be exactly two dashes followed by a space on a line alone to indicate the end of the mail content. But not all mail readers will honor it. -- -Bob Beer

Re: [asterisk-users] Mailing list question 2

On 20 Jan 2011, at 17:13, Andrew Thomas wrote: > Sorry about this - testing this disclaimer problem :) I can give you a POP3 account on my server if it stops you spamming the list?.. S -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Mailing list question

> Is the any kind of 'tag' that I can include at the end of my message > to make the list processing software ignore and dispose of my > disclaimer? It looks like there were underscores on the same line as the -- I think the actual idea is to include '-- ' with nothing else on that line --Don

[asterisk-users] Mailing list question 2

Sorry about this - testing this disclaimer problem :) -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confid

Re: [asterisk-users] context problem

On 01/20/2011 10:58 AM, Jonas Kellens wrote: [snip] I have the following registrations : register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [snip] Problem

Re: [asterisk-users] Mailing list question

That's my last option Jon. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: 20 January 2011 16:59 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mailing list question On 01

Re: [asterisk-users] Mailing list question

Let's see :) -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 20 January 2011 17:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Mailing lis

Re: [asterisk-users] Mailing list question

On 01/20/2011 12:01 PM, Andrew Thomas wrote: why not just subscribe with an account that doesn't do that like gmail or yahoo ? Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words -

Re: [asterisk-users] Mailing list question

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Thursday, January 20, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mailing list questio

[asterisk-users] Mailing list question

Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added

Re: [asterisk-users] context problem

I always thought the last bit (after the /) is where the context in sip.conf landed. What about: (sip.conf) register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [52525252] ... context = TRUNKin52 ... [59595959] ... context = TRUNKin59 ... And

Re: [asterisk-users] res_fax

On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Appl

Re: [asterisk-users] Accessing a 'user' variable via. dialplan.

That's what I am already using :) Somehow, the outbound ID sometimes gets messed up (maybe to do with 2 calls from different users at once) - and the wrong one is sent to the telco. So, rather than just using a 'Set(CALLERID(num)=callidnum' just before Dial - I wanted to check the user directly (

Re: [asterisk-users] ReceiveFax

From: "William Stillwell" Sent: Thursday, January 20, 2011 11:26 AM This is new to me, I have a fax server using Receive Fax and gets way over 5 calls at a time. [fax-in] exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d

Re: [asterisk-users] ReceiveFax

On 01/20/2011 11:00 PM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to th

Re: [asterisk-users] context problem

On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote: Hi Jonas, What else can I try ? Yeah, Asterisk always assumes that from 1 ip address there can only be inbound number. Not very user-friendly. I think I've used something like this: exten => s,1,Set(CALL-TO=${SIP_HEADER(TO)}) exten =>

Re: [asterisk-users] Accessing a 'user' variable via. dialplan.

> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Andrew Thomas > Sent: Thursday, January 20, 2011 11:26 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Accessi

[asterisk-users] Accessing a 'user' variable via. dialplan.

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201

Re: [asterisk-users] ReceiveFax

This is new to me, I have a fax server using Receive Fax and gets way over 5 calls at a time. [fax-in] exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) ;exten => s,n,Set(${LOCALSTATIONID}) exten => s,n,MixMo

Re: [asterisk-users] context problem

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, January 20, 2011 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] context problem On 01/20/2011 0

Re: [asterisk-users] context problem

Hi Jonas, > What else can I try ? Yeah, Asterisk always assumes that from 1 ip address there can only be inbound number. Not very user-friendly. I think I've used something like this: exten => s,1,Set(CALL-TO=${SIP_HEADER(TO)}) exten => s,n,Set(CALL-FROM=${CALLERIDNUM}) exten => s,n,GotoIf($["$

Re: [asterisk-users] context problem

On 01/20/2011 04:29 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, January 20, 2011 9:20 AM *To:*

Re: [asterisk-users] ReceiveFax

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda Sent: Thursday, January 20, 2011 9:00 AM To: Asterisk Asterisk Subject: [asterisk-users] ReceiveFax Hi all, I realize that the application Receivefax can't

Re: [asterisk-users] OT - TTS in spanish

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, January 20, 2011 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT - TTS in spanish Hi, For an organizat

Re: [asterisk-users] Hi, agent intro-speech for outside caller

On Thu, 20 Jan 2011, Danny Nicholas wrote: All Asterisk prompts are configurable with a little legwork.  Simply use the CLI to see what is playing at the point you want to change, then set up this little ditty to override it.  Say you wanted to record the “canned” tt-weasels prompt (“Weasels h

Re: [asterisk-users] context problem

On 01/20/2011 04:43 PM, Jose P. Espinal wrote: Jonas Kellens wrote: [snip] register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten => _52525252,1,NoOp(context TRUNKin - 52525252) exten => _52525252,n,GoTo(blabla,52525252,1) exten => _5

Re: [asterisk-users] ReceiveFax

Hi, I set up ReceiveFax to answer a specific number (2134-4805) , so , the first caller get the fax signal and transmit the fax normal, but, if another caller to call the same number almost at the same time, it gets the signal as well but the fax is not sent! Att, Flavio Roberto Mira

[asterisk-users] OT - TTS in spanish

Hi, For an organization welcoming turists (in France), I would be curious to learn about successful use (with Asterisk) of Text-To-Speech in spanish (and english). I took a look at Cepstral's web site and saw there 2 "Americas Spanish" voices (along a bunch of english voices). 1. In this context

Re: [asterisk-users] context problem

Jonas Kellens wrote: [snip] register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten => _52525252,1,NoOp(context TRUNKin - 52525252) exten => _52525252,n,GoTo(blabla,52525252,1) exten => _59595959,1,NoOp(context TRUNKin - 59595959) exten

Re: [asterisk-users] context problem

On 01/20/2011 04:29 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, January 20, 2011 9:20 AM *To:*

Re: [asterisk-users] context problem

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, January 20, 2011 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context problem Hello list, Asteri

Re: [asterisk-users] ReceiveFax

On Thu, Jan 20, 2011 at 10:00 AM, Flavio Miranda wrote: > Hi all, >  I realize that the application Receivefax can't handle with more than one > fax at the same time. In a environment  with a lot of fax, some caller get > the signal but the operation can't be completed. >  Is  there a way to send

[asterisk-users] context problem

Hello list, Asterisk 1.6.16.1 I have the following registrations : register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [119909] type=friend host=sip.prov.org username=119909 defaultuser=119909 secret=passwd context=TRUNKin extensions.conf : [TRUNKi

Re: [asterisk-users] ReceiveFax

On 01/20/2011 09:00 AM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the secon

Re: [asterisk-users] res_fax

On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? An

[asterisk-users] ReceiveFax

Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Att, Flavio Roberto Mirand

Re: [asterisk-users] Hi, agent intro-speech for outside caller

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Swedrowski Sent: Thursday, January 20, 2011 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hi, agent intro-speech for outsid

Re: [asterisk-users] Top Posting

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Thursday, January 20, 2011 3:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Top Posting > How

Re: [asterisk-users] Hi, agent intro-speech for outside caller

On 20 January 2011 11:54, Tom Rymes wrote: > I don't know of a way to do that, but I can say that, as a caller, it is > highly annoying. Your agents ought to be able to do that themselves, no? > Exactly, otherwise you are losing first chance to make the call "different" from the other ones where

Re: [asterisk-users] Hi, agent intro-speech for outside caller

On Jan 19, 2011, at 11:08 PM, DSR wrote: > Is there anyway to play prerecorded agent intro-speech (like "Hello, my name > is ") to outside caller when agent picks up? I don't know of a way to do that, but I can say that, as a caller, it is highly annoying. Your agents ought to be able to do

Re: [asterisk-users] Top Posting

> How amusing that you follow that statement by being too lazy to trim > all of the irrelevant crud after your comment by pressing > ctrl-shift-end followed by delete. It works in Outlook. > > Tom This is the problem, everyone has a personal goal. One side wants fast replies at the top, with

Re: [asterisk-users] Asterisk 1.8.2 and digium yum repositories

On Wed, 2011-01-19 at 11:41 -0600, Jason Parker wrote: > On 01/19/2011 04:41 AM, Ishfaq Malik wrote: > > Hi > > > > Does anyone have any idea how long it will take for the new release of > > asterisk 1.8 to make it to the digium yum repositories? > > > > Thanks > > > > Ish > > They've been there s