[asterisk-users] Newbie alert: VoIP hardware
Hello, Please forgive me for i'm not an asterisk user yet. I've done as much research as I can .. and have the following questions. I'm setting up a new office and a home office and i'm shopping for hardware. Office: 2 analog lines Hardware: TDM412B (2 FXO, 1FXO) Link: http://www.voipsupply.com/index.php?cPath=99_555_556 Cost: $303 Home: 1 analog line Hardware: TDM421B (2 FXS, 1 FXO) Link: http://www.voipsupply.com/product_info.php?products_id=3980 Cost: $300 Questions: [1] Can I use oslec for echo cancellation? I'll have beefy hardware. Is echo cancellation necessary? [2] Can I get PCI express x1 cards for the same price? I'm on budget, Any other cards (sangoma? rhino?) that might work well? I'm sure these questions have been asked before.. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie alert: VoIP hardware
> > > > If your budget is tight and you want a decent card (not an X100P) with > room to upgrade, then check out > http://www.openvox.com.cn/products.php?genre_id=25 or > http://store.getvoicecards.com/index.php?cPath=66 they are the > reference design that Digium used on previous cards and are very well > made. You can even use their FXO/FXS modules in a real Digium card > and visa versa. > > Thank you all so much for the information. As much as I like to purchase Digium hardware, I'm on a budget, I like the OpenVox PCIe A400E22 (2 FXO/2FXS) card which has enough room for expansion. http://www.openvox.com.cn/products_detail.php?genre_id=38&id=109 I wil be using a Intel C2D E7200 with Micro-ATX M/B in a showbox case. The M/B has 1 PCIe 16x, 1 PCIe 1x and 2 PCI slots. The PCI slots will be used by TV tuners for MythTV. I could not find any US distributors for Openvox cards and I don't want to pay international shipping. Any ideas? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102-NA firmware upgrade on Linux
On Wed, Aug 20, 2008 at 7:44 AM, Joseph <[EMAIL PROTECTED]> wrote: > > Does anybody know if the process of upgrading firmware on "Linksys > SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org > http://www.voip-info.org/wiki/view/Sipura > I just recently upgraded mine. It's very straight forward doing it over HTTP. Download the latest firmware from Linksys. Connect to your SPA3102 via HTTP http://192.168.0.1 Goto the firmware upgrade page. Browse the firmware you downloaded and hit Upgrade. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird asterisk error: ztscan command not found
Hello,
I've installed Asterisk and Asterisk GUI 2.0. The GUI says "No Analog
Card found" and /etc/asterisk/ztscan.conf is empty.
I see the following message from asterisk,
-- Executing [EMAIL PROTECTED]:1]
System("Local/[EMAIL PROTECTED],2", "uptime >
/var/lib/asterisk/static-http/config/sysinfo_output.html") in new
stack
-- Executing [EMAIL PROTECTED]:2]
Hangup("Local/[EMAIL PROTECTED],2", "") in new
stack
== Spawn extension (asterisk_guitools, executecommand, 2) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
-- Executing [EMAIL PROTECTED]:1]
System("Local/[EMAIL PROTECTED],2", "touch
/etc/asterisk/applyzap.conf") in new stack
-- Executing [EMAIL PROTECTED]:2]
Hangup("Local/[EMAIL PROTECTED],2", "") in new
stack
== Spawn extension (asterisk_guitools, executecommand, 2) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
-- Executing [EMAIL PROTECTED]:1]
System("Local/[EMAIL PROTECTED],2", "ztscan >
/etc/asterisk/ztscan.conf") in new stack
/bin/sh: ztscan: command not found
[Aug 29 00:37:13] WARNING[3699]: app_system.c:107 system_exec_helper:
Unable to execute 'ztscan > /etc/asterisk/ztscan.conf'
== Spawn extension (asterisk_guitools, executecommand, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
== Parsing '/etc/asterisk/ztscan.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
== Parsing '/etc/asterisk/zaptel_guiRead.conf': Found
== Parsing '/etc/asterisk/../zaptel.conf': Found
My ztscan is installed in /sbin/ztscan.
And running /sbin/ztscan produces,
[EMAIL PROTECTED] ~]# /sbin/ztscan
[1]
active=yes
alarms=UNCONFIGURED
description=wrtdm Board 1
name=WRTDM/0
manufacturer=
devicetype=
location=
basechan=1
totchans=24
irq=0
type=analog
port=1,none
port=2,none
port=3,FXO
port=4,FXO
port=5,none
port=6,none
port=7,none
port=8,none
port=9,none
port=10,none
port=11,none
port=12,none
port=13,none
port=14,none
port=15,none
port=16,none
port=17,none
port=18,none
port=19,none
port=20,none
port=21,none
port=22,none
port=23,none
port=24,none
[EMAIL PROTECTED] ~]#
Here's a bit more about my hardware,
PC Hardware:
AMD X2 4600+ AM2 CPU
Gigabyte GA-EM780G-S2H (rev 1.1)
2GB DDR2 800MHz
VoIP Hardware:
Sangoma A200 PCI-E (No HW echo canceller)
2x FXO Ports
Software:
CentOS 5.2 x86_64 up-to-date
Sangoma:
Wanpipe (STABLE) 3.2.7.1
Asterisk:
asterisk 1.4.21.2
libpri 1.4.5
zaptel-1.4.11
asterisk-addons-1.4.7
How do I tell asterisk-gui to find /sbin/ztscan so that it can
populate /sbin/ztscan and hence find the hardware?
Thanks in advance,
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial timeout to cell phones
Hello,
I'm new to asterisk and i'm having a really good time configuring it.
I'd like to VoIP-to-PSTN call my SIP number (${MYSIP}) first and then
my cell phone (123456) and then finally to my voicemail.
Here's my dialplan.
exten => s,1,Answer()
exten => s,n,Dial(${MYSIP},20)
exten => s,n,Dial(Zap/g0/123456,10)
exten => s,n,Voicemail([EMAIL PROTECTED])
exten => s,n,Hangup()
When I receive calls, my IP phone (SIP) rings for 20 seconds as
expected. However, asterisk does not terminate calls to my cell phone
after 10 seconds and keeps ringing.
The call to my cell phone is then answered by cell phone voicemail
instead of asterisk voicemail.
Any ideas how to go about this?
Thanks!
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial timeout to cell phones
>
> It was challenging to figure this out, since a lot of the online
> examples seem to work differently, depending on older versions of Asterisk.
>
> I wanted to ring my cellphone (via SIP provider) and deskphone (via Zap)
> simultaneously, but didn't want the call to end up with the cellphone
> voicemail, so press "1" on my cellphone if I want to accept the call
> there. I even see the original caller ID of the inbound caller on my
> cellphone, since I'm out-dialing via a SIP provider.
>
> [inbound]
> exten => 211212,1,Playtones(ring) ; play fake ring so caller doesn't
> wonder
> exten => 211212,n,Dial(Zap/g10&local/[EMAIL PROTECTED],,) ; ring FXS and
> cell
>
> ; http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
> ;
> [internals]
> exten => 101,1,Dial(${MARKCELL},30,tM(screen)) ; play message before
> connecting
>
> ; http://www.voip-info.org/wiki/view/Asterisk+tips+findme
> ; play message to cellphone before connecting inbound call
> ; http://lists.digium.com/pipermail/asterisk-dev/2005-June/013598.html
> ;
> [macro-screen]
> exten => s,1,Wait(0.5)
> exten => s,n,Read(ACCEPT,inbound,1,,1,20)
> exten => s,n,GotoIf($["${ACCEPT}" = "1"]?yes:no)
> exten => s,n(yes),Background(connecting)
> exten => s,n,Goto(end) ; Continue on in dialplan to bridge the call
> exten => s,n(no),Set(MACRO_RESULT=CONTINUE) ; Hangup the called party and
> continue on in the dialplan
> exten => s,n(end),NoOp
>
Thanks for the detailed response, Mark. Looks like a clever trick!
I'll try this out and post results soon.
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G722 and Asterisk 1.6
I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
>>> I have a Grandstream GXP1200 and eager to try this codec. I've heard >> good things about the quality. >> >> Anyone tried it with asterisk? >> >> I can't until 1.6 is released. > > > I have used G.722 with Asterisk many times. If you have more specific > questions about it and Asterisk, I would be happy to try to answer them. Specifically my questions are, [1] The quality of voice between g722 and say GSM or 729 [2] Interoperability between phones with g722 and other codecs [3] Asterisk support for G722 phones. Thanks! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial out via fxo gateway
On Thu, Sep 4, 2008 at 4:44 PM, ACL <[EMAIL PROTECTED]> wrote:
> My current config:
>
> pstn -> audiocodes fxo gateway -> asterisk -> xlite
>
> every fxo ports are registered with asterisk
>
> I have this extensions.conf
>
> exten => 111,1,answer
> exten => 111,n,dial(sip/fxo1)
> exten => 111,n,hangup
>
> If we dial 111 by xlite, I could hear pstn dialing tone. I could key in a
> phone no and connect to the called party. this is a two stage dialing.
>
> How could we preset a phone no. in the extensions.conf without having the
> sip client keys in the phone no (ONE STAGE DIALING)? I do not want to preset
> the phone no. in fxo gateway. the phone no. must be modifiable.
>
> pls kindly advise.
>
I usually have a simple outbound context
[outbound]
exten => _9X.,1,Dial(Zap/g0/${EXTEN:1})
exten => _9X.,n,Congestion()
exten => _9X.,n,Hangup()
Be warned that the above dialplan will allow calls with anykind of
numbers (even international). So be sure to pattern match depending of
where the calls should go.
Don't forget to include the [outbound] context in whatever context
your SIP extention is in.
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
On Thu, Sep 4, 2008 at 7:57 PM, Peder @ NetworkOblivion
<[EMAIL PROTECTED]> wrote:
>> I'd also be more sold on it if it had half the features of the GXP2000
>> (which is only a little over half the price).
>
> Sure, but if only half of the features in the GXP2000 actually work,
> what is the point of them? I'd take a stable phone with less features
> over one that has lots of features that don't work correctly any day.
> I've opened numerous tickets with Grandstream and their answer is always
> "we will look into it" and they never reply, or "that doesn't work" and
> then no reply.
I agree! I bought a GXP1200 ("business" class phone) and it's buggy.
Can't use the message button (404 not found).. and some other features
(404 not found). I have requested help from Grandstream and so far
nothing.
I don't really think they test important features supported by their
phones just the basic ones (dial out/dial in) and that about it :)
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK supported Video phone
On Thu, Sep 4, 2008 at 1:48 PM, Gordon Henderson <[EMAIL PROTECTED]> wrote: > On Thu, 4 Sep 2008, Tharanga wrote: > >> Hi folks, >> >> Can some one recommend a good video phone for asterisk (SIP.IAX2). I need >> better quality, duarability. and should support various video codec's >> .(Codec auto negotiation support id prefferble) > > I suspect that the choices are so limited right now that "good" or "bad" is > going to be very subjective. Grandstream GXP3000's appear to work from what > I've heard others say, as does X-Lite (eyebeam?) softphone, Ekega > (softphone) and ATL video phones... > > Then theres Polycoms with an extra zero aded to the price... > > Some people have reported good results with the BT Videophone 1000 units > too.. (avalable for <£60 a pair, but they need to have the early s/w release > on them) > > I'm just about to order up a paid of Grandstreams for a project... > > (Hm. Can I trunk video over IAX?) > Dlink has launched one in india. http://www.techgadgets.in/misc-gadgets/2008/13/d-link-gvc-3000-ip-videophone-and-glv-540-ip-phones-announced-in-india/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
On Fri, Sep 5, 2008 at 8:16 PM, Tony Mountifield <[EMAIL PROTECTED]> wrote: > In article <[EMAIL PROTECTED]>, > Russell Bryant <[EMAIL PROTECTED]> wrote: >> >> On Sep 3, 2008, at 8:32 PM, sean darcy wrote: >> >> > Great. >> > >> > But I'm still a little confused. >> > >> > Does zaptel 1.4.12 work with asterisk-1.6.0-rc4? >> >> No. Asterisk 1.6.0 now _only_ supports DAHDI. >> >> > It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We >> > can go back to this release of zaptel if we have problems with dahdi. >> > >> > Or if we go back to zaptel, do we go back to 1.6.0-beta9 also? >> >> You can upgrade directly to DAHDI. However, if you have trouble with >> DAHDI and need to go back to Zaptel, then I would go back to Asterisk >> 1.4 instead of using an old beta of 1.6.0. Many things have been >> fixed since 1.6.0-beta9. > > If I'm installing a new system based on the latest Asterisk 1.4, should > I use zaptel or dahdi with it? Which version? > AFAIK, Zaptel is no longer supported. I'd recommend dahdi if it's a new setup. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_dahdi missing in * 1.6.1.1
Hello, I have a Sangoma A200 analog card with 2 FXO ports. It's working well with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk 1.6/dahdi. I compiled and installed, dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10 wanpipe-3.4.1 asterisk 1.6.1.1 My analog card is recognized in dahdi_hardware. However, asterisk cannot compile chan_dahdi.so. I've tried passing --with-dahdi to dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 src yet no luck. I tried passing --with-dahdi to dahdi-tools install dir yet no luck. What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x? I'm on centos 5.3. Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the previous versions do. What changed or what am i missing? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_dahdi missing in * 1.6.1.1
>> What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x? I'm >> on centos 5.3. >> >> Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the >> previous versions do. What changed or what am i missing? > > There probably isn't magic. If you post the errors you got during the > compile we'll be more likely to be able to tell you what's going > wrong. > > Specifically the stuff you got when you said you cannot compile > chan_dahdi.so would be important to post. Ah, there are no errors as such during compile or atleast I didn't seem to notice any. However, I do not see chan_dahdi.so in lib/asterisk/modules/ after asterisk install. Attached are some info and config.log from asterisk 1.6.1.1.. # /opt/dahditools/sbin/dahdi_hardware pci::03:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card # /opt/dahditools/sbin/dahdi_scan [1] active=yes alarms=UNCONFIGURED description=wrtdm Board 1 name=WRTDM/0 manufacturer= devicetype= location= basechan=1 totchans=24 irq=0 type=analog port=1,none port=2,none port=3,FXO port=4,FXO port=5,none port=6,none port=7,none port=8,none port=9,none port=10,none port=11,none port=12,none port=13,none port=14,none port=15,none port=16,none port=17,none port=18,none port=19,none port=20,none port=21,none port=22,none port=23,none port=24,none # cat /proc/dahdi/1 Span 1: WRTDM/0 "wrtdm Board 1" (MASTER) 1 WRTDM/0/0 2 WRTDM/0/1 3 WRTDM/0/2 4 WRTDM/0/3 5 WRTDM/0/4 6 WRTDM/0/5 7 WRTDM/0/6 8 WRTDM/0/7 9 WRTDM/0/8 10 WRTDM/0/9 11 WRTDM/0/10 12 WRTDM/0/11 13 WRTDM/0/12 14 WRTDM/0/13 15 WRTDM/0/14 16 WRTDM/0/15 17 WRTDM/0/16 18 WRTDM/0/17 19 WRTDM/0/18 20 WRTDM/0/19 21 WRTDM/0/20 22 WRTDM/0/21 23 WRTDM/0/22 24 WRTDM/0/23 I does not matter if i pass --with-dahdi to ./configure script or not. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Grandstream, call pickup, ...
On Tue, Jun 9, 2009 at 4:50 PM, Philipp Kempgen wrote: > Thomas Kenyon schrieb: >> Peder wrote: >>> Decent product, but their support and development are horrible. I showed >>> them that their SIP over TCP implementation was broken and their reply was >>> "use udp" >>> >> Such a shame it sounds like it has gone down hill, previously when I've >> spoken to them the standard response was that they'll pass my comments >> on to the development team. > > The development team a.k.a. /dev/null? ;-) > Pretty much. I have a GXP 1200 which sorta works ok. I hope they will work on their products and customer service. They do have a couple of firmwares in beta which might make things better? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
