Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem

On Tue, 30 Nov 2010, Steve Jones wrote: Just the contrary - Most QoS schemes will drop TCP first, specifically because it is known that with TCP, the packet will be resent, so no application will be hurt.  UDP is not dropped first because it is known that there will likely be more impact. I

Re: [asterisk-users] How to hangup all channels

Un-un-top posting... > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Edwards > Sent: 29 November 2010 15:43 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-u

Re: [asterisk-users] Firewalling and Asterisk

On Sun, 28 Nov 2010, Jeremy Kister wrote: > On 11/28/2010 12:03 PM, Silver Thorne wrote: >> So, I am wondering if anyone has a firewall/IP tables statement that >> keep out unauthorised users? No one seems to get in as we use really > > http://jeremy.kister.net/code/iptables/ > > if you already h

Re: [asterisk-users] Recommended *WRT router to run Asterisk?

On Thu, 18 Nov 2010, John Novack wrote: > AstLinux on various embedded hardware really works well. I have several on > older HP Thin Clients, 55xx series, some > with only 128 Meg of ram. A replacement flash from Transcend and a 55xx of > eBay can be easily under $50. > the GUI in AstLinux mak

Re: [asterisk-users] GSM and SS7 Questions

On Wed, 17 Nov 2010, Cary Fitch wrote: > In regard to #2, "any" T1 card should work. But the problem is you need SS7 > software and SS7 connectivity in addition to the T1 card. > > Cary Fitch > Sangoma has both - the card and the software. I think you have to dedicate a box to SS7 though...

Re: [asterisk-users] T1 with Robbed Bit Signaling

On Tue, 16 Nov 2010, Cary Fitch wrote: > Subject: Re: [asterisk-users] T1 with Robbed Bit Signaling > > How many T1s per server? We have never used more than 2 in one server. We don't have very high traffic levels per location. Multiple people on the list have claimed multi-hundred simultan

Re: [asterisk-users] T1 with Robbed Bit Signaling

ndle whatever flavor they give you. j > > To enhance the question: Is anyone on list getting customer loop EELs from > a US (AT&T) Telco, and if so, what flavor and other details? > > Cary Fitch > > > boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere > >

Re: [asterisk-users] T1 with Robbed Bit Signaling

On Tue, 16 Nov 2010, Cary Fitch wrote: > Has anyone here used T1s with RBS with asterisk? > > Cary Fitch > All of my T1s are RBS. No PRI service here. Am using both Digium and Sangoma cards. j -- _ -- Bandwidth and Colocat

Re: [asterisk-users] "scratchy" sound on TE410P

On Thu, 11 Nov 2010, Russ Meyerriecks wrote: >> >> Yes, this is a snapshot after about 24 hours since I cleared the counters. >> I see what you mean - how can I have 76 seconds of errors but no bumped >> error counters. I ran again just now: >> >> r...@vigw3:/etc/asterisk# dahdi_maint -s 1 >> S

Re: [asterisk-users] "scratchy" sound on TE410P

On Thu, 11 Nov 2010, Russ Meyerriecks wrote: >> On Tue, 9 Nov 2010, Daniel Tryba wrote: > >> I am curious about the tool "dahdi_maint"... what do the various >> acronyms stand for? > Yea there seemed to be a bit of confusion here as well so I patched > trunk with some more descriptive error coun

Re: [asterisk-users] "scratchy" sound on TE410P

On Tue, 9 Nov 2010, Daniel Tryba wrote: > On Mon, Nov 08, 2010 at 02:44:26PM -0500, Jeff LaCoursiere wrote: >>> It could be the echo canceller, I had this kind of problem with OSLEC. I >>> also thought the PRI provider was sending clipped audio. I switched to >>>

Re: [asterisk-users] "scratchy" sound on TE410P

On Mon, 8 Nov 2010, Daniel Tryba wrote: > On Sun, Nov 07, 2010 at 10:26:59AM -0500, Jeff LaCoursiere wrote: >> asterisk 1.4.35 >> dahdi 2.3.0.1+2.3.0 >> one span on a 4port T1 card >> >> Got complaints this morning that outbound and inbound calls were >>

[asterisk-users] "scratchy" sound on TE410P

Sangoma has a cool tool "wanpipemon" that shows error stats and such on the span. Is there such a tool for Digium cards? Any suggestions? Thanks, -- Jeff LaCoursiere SunFone j...@sunfone.com -- _ -- Bandwi

Re: [asterisk-users] FW: Under heavy attack

On Mon, 1 Nov 2010, Zeeshan Zakaria wrote: Hi Cary, Can you email me off the list to point it out? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 1:37 PM, "Cary Fitch" wrote: I was going to point out a failing of the attackers, but figured th

Re: [asterisk-users] Under heavy attack

On Sat, 30 Oct 2010, Joel Maslak wrote: > > For me, monitoring outbound call volume makes a lot more sense. I would > love to see an easy to use, out of the box method to alert me if more > than "x" number of erlangs* are exceeded within a five minute, sixty > minute, and one day time period

Re: [asterisk-users] SIP Blacklisting

On Thu, 21 Oct 2010, Steve Howes wrote: > > On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote: >> I'll subscribe, that is for sure. What is the best way to dist the >> blacklist? iptables include file? Or something more integrated to >> asterisk... just thinking

Re: [asterisk-users] SIP Blacklisting

On Thu, 21 Oct 2010, Andrew Latham wrote: > Always start here... http://www.spamhaus.org/drop/ > > If the AS is stolen, you can block the network and never have to worry > about it... > > > ~ > Andrew "lathama" Latham > lath...@gmail.com > I guess you are assuming that spam networks should be i

Re: [asterisk-users] Asterisk Realtime Billing Question???

[snipped very confusing top and bottom posting mix] On Thu, 21 Oct 2010, Sherwood McGowan wrote: > Dhaval, > > You're right, I forgot one thing. The "frozen" table's id column should not > be an autoincrement, it should be set by the insert statement, using the > original method I decsribed for

Re: [asterisk-users] SIP Blacklisting

On Thu, 21 Oct 2010, Steve Howes wrote: > Hi, > > Given the recent increase in SIP brute force attacks, I've had a little > idea. > > The standard scripts that block after X attempts work well to prevent > you actually being compromised, but once you've been 'found' then the > attempts seem to

Re: [asterisk-users] fraud advice

On Fri, 2010-10-15 at 11:20 -0400, Steve Totaro wrote: > This is nothing new. Trunk to trunk transfers and other exploits > could be used on old school phone systems to do the same thing. > > I would start with getting the current balance, if over $10k call the > FBI, call them anyways, it could

[asterisk-users] fraud advice

Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was "taken advantage of" over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern

Re: [asterisk-users] looking for a better ATA

On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Ha

Re: [asterisk-users] random hangups on RBS T1

On Tue, 21 Sep 2010, Shaun Ruffell wrote: > On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote: >> I have several servers with Sangoma A104d cards, and the Sangoma driver >> has a debug mode that lets me see the RBS bit transitions. I have used >> this in the past to prove th

[asterisk-users] random hangups on RBS T1

Hi, I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four port T1 card. Only one RBS T1 plugged into it right now. I have been getting complaints about random hangups. Endpoints are all remote, but I track very closely the latency (by graphing the output of "sip show peers") whic

Re: [asterisk-users] SIP 800 Origination/Termination - International

On Wed, 15 Sep 2010, Kyle Kienapfel wrote: On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere wrote: On Tue, 14 Sep 2010, Joe Freeman wrote: > Anyone have a good provider for International (US/Canada at least) 800 > termination/origination? I have a customer that

Re: [asterisk-users] SIP 800 Origination/Termination - International

On Tue, 14 Sep 2010, Joe Freeman wrote: > Anyone have a good provider for International (US/Canada at least) 800 > termination/origination? I have a customer that had us port one of their > 800 numbers and apparently didn't realize that they had published that > number in Canada as well. Our curr

Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

[snip] customers, who all connect from behind their home nat gateways of all kinds.  I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting

Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

> -- > www.ilovetovoip.com > > > On 2010-09-11 7:22 PM, "Paul Belanger" > > wrote: > > > > > > > > On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere > > wrote: > > >> Sending to 123.123.12... > > > >

Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote: > This is not elastix or FreePBX forum and asking non-asterisk related > questions here is misusing this mailing list. Allow anonymous sip is > not an asterisk feature. Look in the code in extensions.conf what it > is programmed to do and y

Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

On Fri, 2010-09-10 at 23:07 -0400, bruce bruce wrote: > Hi Everyone, > > > I have a provider whose DID used to come into the box just fine but > recently stopped working. Nothing has been changed on our end. > > > Here is what I get when doing "sip set debug peer PROVIDER": > > > Sending to 1

Re: [asterisk-users] Asterisk, HylaFax and Cardiff

On Mon, 23 Aug 2010, Don Kelly wrote: I’m looking for a way to use our implementation of HylaFax on Asterisk with Cardiff (an old installation of Cardiff document stuff). Is someone doing that? If no one has direct experience, is there a HylaFax client that emulates WinFax print-to-fax?

Re: [asterisk-users] Opensource Speech recognition for Asterisk

On Sun, 22 Aug 2010, Jason Aarons (US) wrote: > I'm not aware of an open source speech product. > > Some great examples where speech recognition works well are > 1-800-USA-RAIL, Microsoft/Cisco corporate directory 425-882-8080 you can > say the employees name and be connected and those works g

Re: [asterisk-users] Opensource Speech recognition for Asterisk

On Sun, 22 Aug 2010, David Backeberg wrote: > On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull > wrote: >> Voice recognition is a pain for people with accents and poor lines and when > > Everybody has an accent. Some people live in a place where the people > they talk to sound like themselves,

Re: [asterisk-users] What do you use for Invoicing?

On Tue, 2010-08-03 at 09:17 -0400, bruce bruce wrote: > I agree but the mentioned software is not opensource. > My conditions clearly included opensource. > No, your "prefer" listed "opensource". If you had said "requirement" I wouldn't have suggested it. j > On Tue, Aug 3, 2010 at 12:35 AM,

Re: [asterisk-users] What do you use for Invoicing?

On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote: > Hi Everyone, > > > Sorry, if it's not directly related to Asterisk. Some of people on > this list might have PBX deployed for their clients. What software do > you use to invoice them so the invoice looks like a proper telecom > invoice mayb

Re: [asterisk-users] Video IVR Asterisk ?

On Sat, 2010-07-17 at 00:08 +0530, Anita Hall wrote: > Hi > > Is it possible to receive video calls using Asterisk and then process > them as an IVR ? One of our clients wants to set-up a video IVR system > in the US and we are evaluation possible options. > > Also, what is the bandwidth of rece

Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?

On Tue, 6 Jul 2010, C.Savinovich wrote: > I am writing to you privately... [snip] Doh! Need another cup of coffee? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] call file question

On Wed, 30 Jun 2010, Steve Edwards wrote: >> Now I whipped up a C program to create a call file to do the same thing >> from the command line: >> >> [snip] >> fprintf(callfile, "Channel: Local/*...@custom-callfwd/n\n"); > > I don't see exten "*71" in custom-callfwd. Doh! That was the pr

[asterisk-users] call file question

I am sure this is simple, but have been struggling. I want to create a call file that dials out a particular Dahdi channel to enable call forwarding on a POTS line. I have this in extensions.conf: [custom-callfwd] exten => s,1,Answer exten => s,n,Dial(DAHDI/4-1/*717157750) exten => s,n,Verbos

Re: [asterisk-users] one for your filters

On Wed, 23 Jun 2010, Steve Edwards wrote: > On Wed, 23 Jun 2010, Jeff LaCoursiere wrote: > >> Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place >> four thousand calls to what appears to be a toll number in Zimbabwe last >> night. Filter 82.150.1

Re: [asterisk-users] one for your filters

On Wed, 23 Jun 2010, Tarek Sawah wrote: > > you can start by simply telling us what is the purpose of your server.. > and does it have long distance of overseas?? do you use Numeric > usernames? simple passwords? passwords the same as your username? this > way you can offer more info so we ca

Re: [asterisk-users] one for your filters

On Wed, 23 Jun 2010, Gordon Henderson wrote: > On Wed, 23 Jun 2010, Jeff LaCoursiere wrote: > >> Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place >> four thousand calls to what appears to be a toll number in Zimbabwe last >> night. Filter 82

[asterisk-users] one for your filters

Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll

Re: [asterisk-users] Sangoma - how to show channels in use?

about the individual channels - that is totally handled by asterisk. So it seems there is no way other than what I am already doing to judge the channels in use? Thanks, j > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...

[asterisk-users] Sangoma - how to show channels in use?

Hi, I have several 1.4.29 installations with Sangoma AFT101d cards. Normally we have been collecting the raw data and then graphing channel use for these customers with: asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l Then I recently noticed that there were some "zo

Re: [asterisk-users] Small PC to build and run Asterisk

On Wed, 16 Jun 2010, Randy R wrote: On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere wrote: pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users

Re: [asterisk-users] Small PC to build and run Asterisk

On Wed, 16 Jun 2010, Randy R wrote: > On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves wrote: >> Some distro's, like Askozia and Astlinux, have been specifically >> engineered around running from flash media. This basic form of >> operation has been well proven in projects like monowall and pfsen

Re: [asterisk-users] Dual Atom mobo - call capacity

I don't know how it calculates it, but FreePBX shows a bar for "total calls" that looks like it maxes out at six. We haven't hit that on any installs of this device yet, but that seems pretty low for sure. I know with four calls in progress, all VoIP, transcoding G711u to G.729, the load of t

Re: [asterisk-users] tuning software echo cancellation

On Thu, 10 Jun 2010, Gordon Henderson wrote: > On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: > >> >> On Thu, 10 Jun 2010, Gordon Henderson wrote: >> >>> On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: >>> >>>> We are totally out of touch on the

Re: [asterisk-users] tuning software echo cancellation

On Thu, 10 Jun 2010, Gordon Henderson wrote: > On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: > >> We are totally out of touch on the subject of software echo cancellation in >> asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand >> that >>

[asterisk-users] tuning software echo cancellation

We have been distributing asterisk servers for several years now, and early on decided that hardware echo can was the way to go. Our first few boxes without it had horrid echo problems, and attempts at tuning in 2006 didn't make any difference. We installed a new server yesterday at a client'

Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)

On Tue, 1 Jun 2010, Mike wrote: > Thanks Joe, > > They are on different segments. Those two NICs share nothing but the > server. > > But more to the point, it doesn't explain why a simple routing rule matching > the destination by IP address works wonderfully, but not one where I match a > fwm

Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration

On Thu, 27 May 2010, Mike wrote: > Hi, > > > > I have a test server with 2 NICs, each with it own IP address. Let`s say > 192.168.1.2 and 192.168.1.3. I would like some phones to register by using > 192.168.1.2 and some by using 192.168.1.3 as the address. > > > > Since the default IP is 192.16

Re: [asterisk-users] "ring splash"

card wait X rings or seconds before answering, that would be great. I'm coming up zero on searches. Its already set to wait for callerid, so I am a bit confused why it is picking up on a splash... seems it should wait for that "second" ring anyway. Cheers, j > > On 5/26

Re: [asterisk-users] VoIP over virtualized VPN

I have several Atom based boxes running OpenVPN and processing up to six simultaneous calls over it with no issues. I am quite sure it could do more. Load is still at .2 :) j On Wed, 26 May 2010, Andrew Hakman wrote: I use openvpn for VOIP traffic all the time. It's not a commercial appl

[asterisk-users] "ring splash"

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the mai

Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?

On Thu, 20 May 2010, Gordon Henderson wrote: > On Thu, 20 May 2010, SIP wrote: > >> Even IF you could get a keyboard with lights you could individually turn >> on and off (never seen one), > > http://www.artlebedev.com/everything/optimus/ > > Bit expensive though... > > Gordon > Heh. A $2400 k

Re: [asterisk-users] Digits and Vestec

I'm pretty sure you want it to say "naught" to make the british happy, for zero anyway... j On Tue, 11 May 2010, David Backeberg wrote: Make it say 'zed'. It will make the British happy, and cause a different kind of confusion for the Americans. On Tue, May 11, 2010 at 4:09 PM, Richard Ken

Re: [asterisk-users] OT: NAT in SPA922

On Thu, 6 May 2010, Sebastian Milioto wrote: > I see the following in SPA922 System tab (new firmware) > > VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest > Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID: > VLAN ID:1 for all Phones, and VL

Re: [asterisk-users] OT: NAT in SPA922

On Thu, 6 May 2010, Sebastian Milioto wrote: > It is a building, with 24 separated rooms, each room will have a PC and a IP > Phone. Every room connected to a switch Cisco 2950. > I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), > and still keep communication in same LA

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

On Thu, 29 Apr 2010, David Backeberg wrote: > I'm considering a situation where I buy about twenty ATA devices. > > I've played with the Linksys / Cisco PAP2T, and got that working fine > with some inbound and outbound faxing. The web GUI was okay. I'm > seeing prices around $45 to $50 for this t

Re: [asterisk-users] Testing a sip call through Asterisk?

On Fri, 16 Apr 2010, Nathan Clemons wrote: > I'm looking to find a test tool that will register with our Asterisk > (Trixbox) server here at work and place an outgoing call via our main SIP > trunk (BroadVoice) to confirm that things are working. I've looked around > but I can't seem to find any

Re: [asterisk-users] How can I record the conversations in a conference call?

On Fri, 16 Apr 2010, Carlos Chavez wrote: > On Fri, 2010-04-16 at 08:38 -0700, Luki wrote: >>> "Please note: A Zaptel timer must be present for conferencing to work!", but >>> if the user does not have "ZAP/DAHDI hardware", he can use ZAP/DAHDI DUMMY >> >> Actually, my understanding is that this

Re: [asterisk-users] jitterbuffer

On Thu, 8 Apr 2010, Tim Nelson wrote: > - "Jeff LaCoursiere" wrote: >> What is the consensus on using the 1.4 jitterbuffer? Do most people >> enable it? >> >> We have a "PSTN" server that has our RBS T1 trunks in a central >> location

[asterisk-users] jitterbuffer

What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a "PSTN" server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are h

Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls

On Thu, 8 Apr 2010, bruce bruce wrote: > I am not sure if unplugging line from card would work as it's still in a > hunt and calls will keep coming through that number and won't fall over to > next line unless there is a BUSY on the line. There is no timeout; it's a > hunt on BUSY. Plus, I don't

[asterisk-users] realtime jitter/latency measurements

Howdy, Can anyone point me to links or discussions about realtime jitter measurement? I read a long thread from 2007 (Douglas Garstang) that didn't end with any conclusions. I want to do the same thing he was trying to do - allow realtime jitter measurements to help control call routing with mu

Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

I finally got it at "Calories Consumed". Geesh. Good one! :) j On Thu, 1 Apr 2010, Olle E. Johansson wrote: > FOR IMMEDIATE RELEASE > Puerto Escondido, Mexico, April 1st, 2010: > > Digium launches Asterisk VCC (TM) - a new virtual communication platform > for enterprises, the public sector a

Re: [asterisk-users] 8Port Junghanns BRI card under Dahdi

On Sat, 20 Mar 2010, Loic Didelot wrote: > Hi, > I try to get an 8 Port Junghanns BRI card working under dahdi. The card > works with zaptel but I have no success under dahdi. > > I load the module with "modprobe wcb4xxp". I dont get any errors but I > dont see the spans in /proc/dahdi. The outpu

Re: [asterisk-users] Time counting while playback

On Tue, 16 Mar 2010, Pham Quy wrote: > Hi all, > > This question has been asked for days, I think that would be more > comprehensible if i post it in a new thread. > > What i want to do is something like karaoke. when users call to > asterisk, a music song is played while caller sings. Their voic

Re: [asterisk-users] Asterisk to be used with Ciscs media gateways

> From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere > Sent: Monday, March 15, 2010 6:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk to be used with Ciscs med

Re: [asterisk-users] Asterisk to be used with Ciscs media gateways

On Mon, 15 Mar 2010, David Backeberg wrote: > >> and also to do LCR and Quality based routing of International calls? > > I don't know what that means. > Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that

Re: [asterisk-users] Android Phones ;-)

On Mon, 15 Mar 2010, Ishfaq Malik wrote: > Conrad Wood wrote: >> FWIW, just received an android-based phone and after installing >> "sipdroid" found that it works very well with asterisk ;). >> >> It automatically dials numbers through asterisk if available and >> otherwise through the gsm netwo

Re: [asterisk-users] IAX2 peer question

On Sat, 13 Mar 2010, Jeff LaCoursiere wrote: > > What does the (T) mean? Am playing around with running an IAX trunk over > an OpenVPN session and see this only on this peer. > > demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK > (26 ms) > > Same

[asterisk-users] IAX2 peer question

What does the (T) mean? Am playing around with running an IAX trunk over an OpenVPN session and see this only on this peer. demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK (26 ms) Same thing on the other side: sunfone/demopbx 10.222.0.1 (S) 255.255.255.255 456

Re: [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here

On Fri, 12 Mar 2010, Angelito Manansala wrote: > If you are having trouble reading this email, read the online > version > . > > > >

Re: [asterisk-users] 30 mins GSM file

On Thu, 4 Mar 2010, Steve Howes wrote: > > On 4 Mar 2010, at 23:11, Steve Edwards wrote: >> On Thu, 4 Mar 2010, Steve Edwards wrote: >>> On Fri, 5 Mar 2010, David @ULC wrote: >>> I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way

Re: [asterisk-users] time/date over POTS?

On Thu, 4 Mar 2010, Dave Fullerton wrote: > Jeff LaCoursiere wrote: >> I had a customer ask me about time/date information being sent to his >> analog (attached to a Linksys SPA2102) answering machine. I didn't know >> that POTS could carry this information. Is thi

[asterisk-users] time/date over POTS?

I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j -- __

Re: [asterisk-users] video voicemail

On Mon, 15 Feb 2010, Tilghman Lesher wrote: > On Monday 15 February 2010 14:09:38 Olle E. Johansson wrote: >> 15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere: >>> Playing around with the Grandstream GXV3140. >>> >>> I'm interested in having the video voi

[asterisk-users] video voicemail

Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by Windows Media Player or even Quicktime. Have been googling around a lot and have tried various bits of OSS to read the resulting .h264 file that asterisk

Re: [asterisk-users] IP Phone recommendation

On Wed, 10 Feb 2010, Tim Nelson wrote: > - "Gordon Henderson" wrote: >> If not using PoE I'd suggest getting a few extra PSUs though - that's >> one >> area I have had a few issues with - but maybe it's just been the UK >> ones. >> >> Gordon > > The same can be said for the US versions. My e

Re: [asterisk-users] test

fail. On Tue, 9 Feb 2010, aster...@opensourcesolution.in wrote: > > > test -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Dial script

On Sat, 6 Feb 2010, Thomas Perron wrote: karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife wrote: Try this: #rm -rf / I second that opinion. Tell us first WHY you want to dial 1 numbers "in sequence". Without any reason, you must be assum

Re: [asterisk-users] large scale paging

On Fri, 5 Feb 2010, Mark Willis wrote: > Has anyone done any large scale intercom deployments with Asterisk? I've > been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP

Re: [asterisk-users] Losing local SIP phones when internet goes down?

On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line "nameserver 127.0.0.1" on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. j Atenciosament

Re: [asterisk-users] Losing local SIP phones when internet goes down?

On Fri, 5 Feb 2010, Nikhil Nair wrote: > Hi again, > > OK, I've now installed a local caching nameserver, but don't see any > change at all. > > IN detail, what I did: > > - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes > care of dynamic nameserver allocations in /etc/re

[asterisk-users] asterisk video support and IPTV

Has anyone played with the idea of Asterisk as an H.264 multicast tool? I am wondering what the possibility would be to have some kind of machine with a capture card call asterisk over SIP and have asterisk make another hundred calls to subscribers. Then any H.264 compatible device (Android?

Re: [asterisk-users] codec conversion

On Tue, 2 Feb 2010, Steve Edwards wrote: > On Tue, 2 Feb 2010, wassim darwich wrote: > >> Thanks for?your reply,ill give?you my situation, iam using my asterisk box >> as a switch ,so my client is sending me ulaw and my voip provider?only >> accept g723 ,So what i have to do is to receive?g711

Re: [asterisk-users] Mitel integration

> On 1/27/2010 11:00 AM, Steve Howes said: >> On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: >>> Sounds good to me, but without the spec I'm stuck in a catch 22! >> >> tcpdump? (assuming IP).

[asterisk-users] Mitel integration

Hi, A potential client (hotel) has a Property Management System that talks the "Mitel" protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or be

Re: [asterisk-users] Virtual Asterisk Installation

On Thu, 21 Jan 2010, Gergo Csibra wrote: > Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote: > >> Forget about virtualization! > ... >> Virtualisation is nice for test-setups, but thats it. for any real job >> it's a major pain in the ass and makes stuff bork beyond imagination. > > Well.

Re: [asterisk-users] 10/100 voip phones and gigabit connection

On Fri, 15 Jan 2010, Hans Witvliet wrote: > > If you connect your pc with GB-lan card to an dual-ported ip-phone, you > and up with an 100Mbps lan connection to your pc. > > Only way to avoid that, is to insert a cheap second lan-card in your pc, > and connect your phone to the second lan, so yo

Re: [asterisk-users] 10/100 voip phones and gigabit connection

On Fri, 15 Jan 2010, randall wrote: >> Sure. My point was just that IF you only got one connection in the wall, >> its cheaper to get a switch than getting a phone with dual 1Gbit ports. >> >> Leif >> > OK, point taken. > > but i have 6xisdn2 and already 2x24 gigabit switches (will need to re

Re: [asterisk-users] GXV3140 and Xlite video

On Thu, 14 Jan 2010, Julian Lyndon-Smith wrote: > Has anyone managed to get these two phones to make a video call to each other > ? > > If so, care to share how the hell you managed ? > > the GXV is at the latest firmware, and xlite the latest download > > Asterisk 1.4 trunk > > TIA > > Julian >

[asterisk-users] Dahdi and FreePBX

Perhaps this more belongs on the FreePBX list, but for the archives, this is what I did to make it work: chan_dahdi wants to read /etc/asterisk/chan_dahdi.conf FreePBX, at least how I installed from source, seems to think I am still running Zaptel. It created zapata_additional.conf when I add

Re: [asterisk-users] Dahdi issues

igium.com] On Behalf Of Jeff > LaCoursiere > Sent: Thursday, January 14, 2010 9:23 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Dahdi issues > > > Hello, > > My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port > modular

[asterisk-users] Dahdi issues

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: r...@pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx :04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 7

Re: [asterisk-users] iaxmodem / hylafax receive problem

On Thu, 14 Jan 2010, Doug Lytle wrote: > Kingsley Tart wrote: >> Hi, >> >> I'm trying to receive faxes using hylafax / iaxmodem but I just can't >> get it to work. We're using Sangoma E1 cards and have calls coming in >> > > Without seeing your config files for iaxmodem and hylafax and also > see

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

On Tue, 12 Jan 2010, Richard Kenner wrote: >> Your comments both come from having taken a short look at FreePBX and >> dismissed it without investigating how powerful it can be. > > Yes, but the discussion is about COMPLEXITY, not power! I thought the discussion was about how an IT guy with no

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