[asterisk-users] e164 Format Numbers

This is probably a very simple question, but I can't for the life of me work it out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the + in the SIP URI, it doesn't

[asterisk-users] Best phone for easy provisioning

individual config is not an option? Rod Bacon Technical Manager JASCO Consulting Pty. Ltd. http://www.jasco.net.au http://www.jasco.net.au/ Ph. 03 9432 6376 Fax: 03 9432 6378

[Asterisk-Users] AGI and Video

or documentation on any of the above? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600    Fax: +613 99401650 FWD: 512237                   ICQ: 5662270

Re: [Asterisk-Users] grandstream GXV-3000

/asterisk-users -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600    Fax: +613 99401650 FWD: 512237                   ICQ: 5662270

[Asterisk-Users] Dialogic Hardware

In case anyone is interested, I have a Dialogic D/600JCT-2E1-120 that we paid about A$15K for not so long ago. I am open to any serious offers. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205

[Asterisk-Users] Simple windows / web Asterisk user software?

Our windows users are looking for a simple application to permit dialling and transfer from Windows desktop (or web page). I've looked at everything mentioned in the WIKI, and most are either not appropriate, or are not maintained any longer. I've used Flash Operator Panel, and quite like it,

Re: [Asterisk-Users] Simple windows / web Asterisk user software?

I liked the look of it, but the documentation didn't mention transfer capability. Does it do transfers? On Tuesday 30 May 2006 10:27, Paul Hales wrote: Have you given SNAP a go? http://www.snapanumber.com/Home/tabid/53/Default.aspx ___ --Bandwidth

[Asterisk-Users] SNOM 190 Daylight Savings

I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne

Re: [Asterisk-Users] Distinctive ring?

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600    Fax: +613 99401650 FWD: 512237

[Asterisk-Users] Grandstream NTP

tried latest (and BETA firmware). Does anyone have any ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600    Fax: +613 99401650 FWD: 512237                   ICQ: 5662270

Re: [Asterisk-Users] Grandstream NTP

It now appears to be server specific. The shipped default, time.nist.gov, appears to work OK. Does anyone know of anything specific required by these grandstream phones as far as NTP server support goes? On Tue, 6 Dec 2005 10:34 am, Rod Bacon wrote: All my BT101's and GXP2000's are failing

Re: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

be because of the different default echo canceller in the Zaptel drivers? Anyway... good luck. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

Re: [Asterisk-Users] DSL router with QOS

. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Keith Schmidt wrote: Any recommendations on an ADSL router

[Asterisk-Users] Changes from 1.2beta2 to 1.2RC-1

I've spent some time clicking my way around the digium website, but can't seem to locate a list of changes from * 1.2beta2 to 1.2RC-1. Can anyone point me in the right direction? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South

Re: [Asterisk-Users] DTMF detection in TE406P ??

Kevin P. Fleming wrote: There are other steps that can be taken if necessary first. Can you please elaborate on this? It may just save a lot of calls to Digium support about the same issue. (I have noticed this sporadically). ___ --Bandwidth and

Re: [Asterisk-Users] HDLC errors on PRI

zaptel, the card drivers and asterisk each time you make a change. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ

Re: [Asterisk-Users] ATA-488 FXO

I tried this unsuccessfully with an early (pre-release) version of the 488 firmware. I haven't tried it recently though. I'll have a play later in the week and let you know... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Digium Boards

. You can also disable stuff like USB, Parallel, Audio, secondary IDE, etc. etc, which can all free-up IRQs. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

Re: [Asterisk-Users] Missing audio from Zaptel channels - SOLVED!?

For those who are interested, the problem appears to NOT exist in 1.2Beta2. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

them out) then when they return, dial another number to do the reverse. Then simply route the calls to the queue using a ringall method. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613

[Asterisk-Users] Re: [Asterisk-biz] New asterisk web gui for small/medium sized businesses

You asked for feedback, so here goes. Let me start by saying that I applaud your effort at getting involved in the project. I wish i could write a scrap of code, as there's literally dozens of things I'd like to contribute. Now... onto AWG. Personally, I'm not sure where this tool fits.

[Asterisk-Users] Missing audio from Zaptel channels

and ideas. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270

Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS

Thanks for the suggestion, but in my experience fax machines on ATAs can yield unpredictable results, even at LAN speeds and uncompressed codecs. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205

Re: [Asterisk-Users] Echo canceller on TE406 Asterisk

I have similar problems with performance degradation over time. I'm about to post another message to the list (once I have some more information). Stay tuned. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria

[Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS

Does anyone out there have any TDM400 FXS module(s) that they want to swap for FXO (preferably in Australia). I have a quad-port FXO arrangement at the moment, but I need to plug a couple of fax machines into my * box... -- == Rod Bacon Empowered

[Asterisk-Users] OT: Samsung DCS 70

port. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270

Re: [Asterisk-Users] Routing landline calls to asterisk.

Short answer: No, but... Long answer: Yes, and... Essentially, there are *certain* internal modems that will handle this function, but basically what you're talking about is an FXO card. You can pick up one for little outlay on eBay. Do a search on eBay for X100P. Then read the wiki for

[Asterisk-Users] Wrong caller id in CDR

, they are being written with the CLID of the _last_ caller to access the specific ZAP channel in question, not the current one. Has anyone ever seen this before? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria

Re: [Asterisk-Users] PRI echo issues: solvable?

I'd be interested to know if this gets worse over time. Shutdown asterisk, remove card driver, load card driver, load asterisk then test. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205

Re: [Asterisk-Users] enable mysql in asterisk

If you want my opinion, a single server (or even a small farm) is still easir to manage with conf files. A simple reload in the * CLI, and you're done. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia

Re: [Asterisk-Users] Wrong caller id in CDR

? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == C F wrote: Yeah, meaning

Re: [Asterisk-Users] Wrong caller id in CDR

as part of a deadAgi. (I already tried to set the callerid to in a deadAgi, it didn't work) == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

Re: [Asterisk-Users] Problems with Wait SIP 486 DND

Or explained more clearly The fallback rule is n + 101, so your voicemail busy priority needs to be 103 (2 + 101). == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600

[Asterisk-Users] Zaptel Line Build Out

: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia

Re: [Asterisk-Users] Zaptel Line Build Out

? asterisk wrote: http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211613,00.html - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 09, 2005 6:42 PM Subject: [Asterisk-Users] Zaptel Line Build Out Can someone who

Re: [Asterisk-Users] Zaptel Line Build Out

, what is meant by DSX-1? What is CSU? Why would I use a -7.5db, -15db or -22.5db LBO? asterisk wrote: http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211613,00.html - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent

Re: [Asterisk-Users] Delay before dialplan is launched?

For those who are interested... I set overlapdial=no, and things are all good again. D'oh! == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

[Asterisk-Users] Latency on bridged PRI calls

Nobody has been able to answer this. Not even Digium at this stage, but I'm hoping someone here, smarter than I, will be able to. We are running some TE406P (upgraded 405Ps) cards performing mainly PRI bridged calls. After a server is brought up, calls sound absolutely perfect. Over time,

Re: [Asterisk-Users] Latency on bridged PRI calls

Upon closer inspection, I don't think my system ever tries to establish a zaptel native bridge. Is there somewhere where this function is enabled/disabled? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria

[Asterisk-Users] Clearing Caller-ID from Zaptel Channels

, the call, at least from a cdr perspective, seems to inherit the identity of the previous caller. Is ths a bug, or by-design? Can this behaviour be modified? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia

Re: [Asterisk-Users] Clearing Caller-ID from Zaptel Channels

No. I'm running 1.2beta from Digium tarballs. Can you point me in the right direction? (the old thread?) == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

[Asterisk-Users] Delay before dialplan is launched?

(approximately) between the time the call is received and the first dialplan command is executed. Is everyone else experiencing this? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600

[Asterisk-Users] Digium hardware echo canceller, zapata.conf settings?

Do the echo cancellation settings in zapata.conf have any effect when hardware echo cancellation is installed on a 406p/411p? How can I tell if the echo is being cancelled by hardware or software? -- == Rod Bacon Empowered Communications Ground Floor

Re: [Asterisk-Users] IBM tts engine integration

Not bad.. but still not as good as Scansoft's... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270

Re: [Asterisk-Users] success story: TE406P (quadspan with hardware echocan)

Which version of asterisk and zaptel are you using? Will they work with 1.0.9 ? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

Re: [Asterisk-Users] Echo after running for several days?

zaptel drivers, but i was hoping for a more 'solid' solution. Does anyone know of one? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

Re: [Asterisk-Users] Clicks in audio with TE100P PRI

Which file does the jitterbuffer setting go in, zaptel or zapata.conf? I can't find it documented anywhere. What version of zaptel drivers include a jitterbuffer? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne

Re: [Asterisk-Users] TE405P V2 changes?

Did you find a solution to this? Kib Eki wrote: yes, fedora 3 but without any changes at the sources Master Abi wrote: Are you using Redhat/Fedora? If I remember those init scripts is for Redhat/Fedora. I am using gentoo. Did you make any modifications to wct4xxp.c. or pass any parameters

[Asterisk-Users] TE405P V2 - Fantastic!

zaptel drivers. The card didn't work with 1.0.9.1 drivers! I do NOT have the echo canceller module installed, as 90% of my calls are zaptel bridged calls. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria

Re: [Asterisk-Users] OT: Online TTS engines?

Can I get a copy of that PERL script? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270

Re: [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!

An update on this... I was wrong. The wireless problem was an altogether different issue. the wj0011 firmware finally made my phone useable, after 6 months of problems. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne

Re: [Asterisk-Users] monitor peak channel use

servers. It also writes RRDs... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270

Re: [Asterisk-Users] What have I misconfigured?

Why is each phone registering twice (2 different ports)? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270

[Asterisk-Users] 410P upgrade to 411P?

Does anyone know if the echo cancellation module can be retro-fitted to a 410P to turn it into a 411P? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

[Asterisk-Users] IAX Trunking Weirdness

I am having trouble getting trunking to work between a couple of my servers. All servers are running 1.0.9 stable version, and are working perfectly. All have a Zaptel card of some description, so timing is not a problem. Each server has a definition for each other server, using RSA auth, and

Re: [Asterisk-Users] IAX Trunking Weirdness

the originating (incorrect) address. Apparently there is a patch against CVS HEAD which allows the definition of the source address on a per-peer basis. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia

Re: [Asterisk-Users] 410P upgrade to 411P?

Did it make a lot of difference? Is the canceller effective? How much CPU will I save by doing it in HW? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

[Asterisk-Users] PRI in and out

I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI - Asterisk side to

Re: [Asterisk-Users] PRI in and out

It DOES help, thanks. Except for this the only difference between the first set of channels (1-23) and the second set of channels (25-47) is: signalling=pri_net group=1 context = fromprovider channel = 1-23 signalling = pri_cpe group=2 context=fromavaya channel=25-47 I thought the

[Asterisk-Users] Zyxel Prestige 2000W Firmware - EVIL

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St

Re: [Asterisk-Users] How can I use MySQL in the dialplan?

You'll have a much more flexible solution if you keep your MySQL access out of the * dialplan, and put it in AGI. Matthew Boehm wrote: What the hell? NO! show application MySql app_addon_mysql is the name of the module. load app_addon_mysql.so -Matthew Quoting Ronald Wiplinger [EMAIL

[Asterisk-Users] Cisco 7940 - Disappearing Clock - SOLVED

Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com This question is not actually * related, but please don't flame me! Is anyone out there using the 7.4 or 7.5 SIP firmware on their Cisco 79xx phones? I

[Asterisk-Users] Cisco 7940 - Disappearing Clock

This question is not actually * related, but please don't flame me! Is anyone out there using the 7.4 or 7.5 SIP firmware on their Cisco 79xx phones? I have a weird problem where my clock disappears after a period of time, and the only thing that will get it back is a reboot. Has anyone

Re: [Asterisk-Users] problems with compiling asterisk-oh323

I suggest you read the installation documentation again. The error is telling you what the problem is. You don't have pwlib and oh323 source compiled (using make opt only) and sitting in the root/src directory. If it's somewhere else, edit the asterisk-oh323 makefile to reflect the correct

Re: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary, D-channel of span 1

2 - Check your span line in your zaptel.conf. You should be receiving timing, not giving it, when using a PRI (generally). Change the second number from 1 to 0. Save and restart asterisk. (span=1,0,0,esf,b8zs) I think you've got this cocked-up. A 0 in the second position tells zaptel

Re: [Asterisk-Users] DELL 2800 : PCI Parity error

systems now run close to 100% in zttest, never miss an irq and don't seem to generate PCI parity errors any more. I don't know if I've fixed it, but you should really go through the whole process anyway. == Rod Bacon Empowered Communications Ground Floor

Re: [Asterisk-Users] sound files

If you do a make install samples in the asterisk src dir, it will put them into /var/lib/asterisk/sounds Chadwick E. Labno wrote: where should the sound (.gsm) files be located? Currently the are in /usr/src/asterisk/sounds. I feel they should be located else ware, like in

Re: [Asterisk-Users] Not MRTG, what about ARGUS?

I use Nagios to monitor lines. I use the check_asterisk script that you'll find floating around the place. I connect via the mgmt interface. Added to nagios is nagiosgraph. This keeps historical RRD graphs of my line usage. == Rod Bacon Empowered

Re: [Asterisk-Users] HDLC bad FCS

Thankyou for an excellent post. Mike M wrote: Comments throughout. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] HDLC bad FCS

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the

Re: [Asterisk-Users] HDLC bad FCS

] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, July 04, 2005 7:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HDLC bad FCS I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server

[Asterisk-Users] No BUSY on PRI

I'm using a TE405P and stable version of Zaptel. When I call a BUSY number on my E1 PRI, I don't get a busy status. The caller hears a busy tone, but the CDR logs a NO ANSWER when the caller hangs up. Is this normal for this version of Zaptel? ___

Re: [Asterisk-Users] Zaptel card AND Ztdummy together?

In case anyone is interested, loading Ztdummy AND a card driver at the same time will result in unpredictable timing issues. We heard intermittent echo/feedback on PRI channels. Rod Bacon wrote: I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10

Re: [Asterisk-Users] Zaptel card AND Ztdummy together?

they have a secondary clock source available across another circuit? Perhaps a tie line to a pbx that can be configured as a secondary? -Original Message- From: Rod Bacon Sent: Thu, June 23, 2005 12:03 am I had a weird (unforeseen) situation today. We have a remote office with an * server

[Asterisk-Users] Zaptel card AND Ztdummy together?

I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX

[Asterisk-Users] Scratchy audio on Bridged PRI Calls

, Frame Buffer, etc. etc.) and am getting good results in zttest. I see NO IRQ misses or any other errors at the console. Does anyone have any other ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia

[Asterisk-Users] Premptible Linux Kernel

Can anyone tell me if Asterisk would speficically benefit from the reduced latency of a preemptible Linux Kernel? I know it was recommended against in the early days, but I'm wondering if there are any updated opinions? -- == Rod Bacon Empowered

[Asterisk-Users] Zaptel HEAD with * Stable?

Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237

[Asterisk-Users] ee1000 Ethernet in Dell 1850

Digium's site now lists the Dell 1850 as a potential problem server, as it uses the intel ee1000 Ethernet chipset (as do a majority of servers in the market!). To my knowledge, ALL dell servers with Gigabit interfaces now use the same chipset. Does this mean the Digium cards can't be used in

Re: [Asterisk-Users] Grandstream ATA Toasted

This is not an option for me, as the IVR menu is nuked as well... Luki wrote: A BETA firmware upgrade toasted my ATA286. It now has limited operations. Happened to me too... looked mostly dead, but not quite. Try a complete hardware reset. See section 8 on

[Asterisk-Users] Grandstream ATA Toasted

A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality.

Re: [Asterisk-Users] ztcfg server crash

:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ztcfg server crash I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel

[Asterisk-Users] OH323 Packetization

Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the number of frames per RTP packet. How does this equate to packetization in ms? ___ Asterisk-Users mailing list

Re: [Asterisk-Users] OH323 Packetization

ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729? Rod Bacon wrote: Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the number of frames per RTP packet. How does this equate

Re: [Asterisk-Users] OH323 Packetization

I answered my own silly question. 10ms. If anyone needs a working OH323 config for Comindico (SPT) in Australia, please mail me (G.729 and G.711). Rod Bacon wrote: ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729? Rod Bacon wrote: Forgive this (possibly

[Asterisk-Users] ztcfg server crash

it seems to be partially screwed. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270

Re: [Asterisk-Users] ztcfg server crash

and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following

[Asterisk-Users] Inject Audio into Existing Call

Other than using a conference, does anyone know of a way to inject audio into a live call between two parties? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server

). * on it's own is reasonably light on resources. Go for it! == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

Re: [Asterisk-Users] CRM integration (was RE: CallerID)

This sounds remarkably like an IM problem We're in the process of building a CRM frontend that uses Jabber as the IM mechanism. The Asterisk server sends the URL via Jabber (PCs authenticated as extension number). The Jabber client (custom, written in Flash) receives the URL and

[Asterisk-Users] Analog Telephone Adapter

An IBM sales rep once told me... I can give you RELIABLE, FAST and CHEAP... any two of them at once. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] International Caller ID?

We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Problems trying to compile H323 from CVS-STABLE

!) -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk

[Asterisk-Users] more than one company hosting their PBX on the same machine?

Sigh... read the wiki. Search the lists. This has been answered at least fifteen times. You don't need multiple instances of *, just set up your dialplan properly. Hint: Contexts are the key. ___ Asterisk-Users mailing list

[Asterisk-Users] Which H.323 for Stable?

I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly

RE: [Asterisk-Users] Static on TDM Zaptel FXO

Make sure you have disabled framebuffer, apic and acpi. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

[Asterisk-Users] ISDN Clock Source

---| Asterisk |---ISDN| Legacy PBX |-- -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

[Asterisk-Users] Database of actve calls (as per astguiclient)

I have the need to maintain a pseudo-realtime database of active calls across a number of asterisk servers. The main purpose of this is in determining where to route calls (e.g. don't send calls to a server with no free lines) and also for monitoring/recirding calls. I know that astguiclient

[Asterisk-Users] TE410P on Dell 2650

. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

[Asterisk-Users] chan_vpb Verbose Logging

Does anyone know if there is a way to turn DOWN the verbosity of the Voicetronix channel driver? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

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