Voxilla VoIP Forum

is there any way to route a PSTN call b4 answer it?

KraYYem — Sun, 08 Nov 2009 23:07:05 GMT

any solutions PLEASE?

BrainTel Down..CAD n Direct both

mirza001 — Sun, 08 Nov 2009 18:32:51 GMT

Made three successful calls to Lahore landlines and mobile numbers, also tried customer service 3210-0786, connecting fine; usually on weekends service is extremely busy so had to call multiple times before calls were connected.

Auto fallthrough, channel 'SIP/davy1-091efb50' status is 'CHANUNAVAI

mehdymahmood — Sun, 08 Nov 2009 08:02:27 GMT

im new in asterisk .
i am using following package
os ubuntu
asterisk-1.6.0.15
dahdi-linux-complete-2.2.0.2+2.20
libpri-1.4.10.1

sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
tcpenable= yes
tcpbindaddr=0.0.0.0
srvlookup= no

register => joahim1:******@192.168.1.101/davy1
register => davy1:******@192.168.1.100/jaohim1

[joahim1]
type=friend
secret=******
qualify= yes
nat=yes
host= 192.168.1.101
dtmfmode=rfc2833
canreinvite=no
username=joahim1
context= mehdy
allow= all

[davy1]
type=friend
secret=******
qualify=yes
nat=yes
host= 192.168.1.100
dtmfmode=rfc2833
canreinvite=no
username=davy1
context=mehdy
allow =all

extension.conf

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2

DAVY=SIP/davy1
TRUNKMSD=1

[mehdy]

exten => 101,1,Answer()
exten => 101,n,Background(demo-nogo)
exten => 101,n,WaitExten()

exten => 1,1,Dial(SIP/jaohn1,10)

exten => 2,1,Dial(${DAVY},10)

ion for 'joahim1@192.168.1.101' timed out, trying again (Attempt #86)
[Nov 8 14:24:14] NOTICE[3656]: chan_sip.c:9740 sip_reg_timeout: -- Registration for 'joahim1@192.168.1.101' timed out, trying again (Attempt #86)
[Nov 8 14:24:34] NOTICE[3656]: chan_sip.c:9740 sip_reg_timeout: -- Registration for 'davy1@192.168.1.100' timed out, trying again (Attempt #87)
[Nov 8 14:24:34] NOTICE[3656]: chan_sip.c:9740 sip_reg_timeout: -- Registration for 'davy1@192.168.1.100' timed out, trying again (Attempt #87)
[Nov 8 14:24:34] NOTICE[3656]: chan_sip.c:9740 sip_reg_timeout: -- Registration for 'joahim1@192.168.1.101' timed out, trying again (Attempt #87)
[Nov 8 14:24:34] NOTICE[3656]: chan_sip.c:9740 sip_reg_timeout: -- Registration for 'joahim1@192.168.1.101' timed out, trying again (Attempt #87)

== Using SIP RTP CoS mark 5
[Nov 8 13:56:51] WARNING[3818]: app_dial.c:1499 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Nov 8 13:56:51] WARNING[3818]: app_dial.c:1499 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/davy1-091efb50' status is 'CHANUNAVAI

please help me .

Polycom IP 330 and Polycom IP 430 reboot Issue

nomykhan — Sun, 08 Nov 2009 07:34:55 GMT

Hi,

The IP Phones installed are Polycom IP 330 and Polycom IP 430. The problem (IP Phones reboot) occurs randomly in any of the phones once a day.

Installed switch is: WS-C3560 48PS-E - Catalyst 3560 48 10/100PoE +4SFP IPS Image

Below is the config of the switch,

enable secret 5 $1$O872$Uwg8SAnBIcvW8U7FN0r01.
!
aaa new-model
aaa group server tacacs+ acs-servers
server 10.21.6.20
server 10.21.6.21
!
aaa authentication login NSG-AAA group acs-servers line
aaa authentication login LINE-AAA line
aaa authentication enable default group acs-servers enable
aaa authorization exec NSG-AAA group acs-servers if-authenticated
aaa accounting exec default start-stop group acs-servers
aaa accounting commands 15 default start-stop group acs-servers
!
aaa session-id common
clock timezone KSA 3
system mtu routing 1500
ip subnet-zero
no ip source-route
no ip domain-lookup
!
!
!
!
no file verify auto
!
spanning-tree mode pvst
spanning-tree loopguard default
spanning-tree portfast bpduguard default
no spanning-tree optimize bpdu transmission
spanning-tree extend system-id
spanning-tree uplinkfast
!
vlan internal allocation policy ascending
!
!
interface Port-channel1
description ***** UPLINK TO CORE *****
switchport trunk encapsulation dot1q
switchport mode trunk
!
interface FastEthernet0/1
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/2
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/3
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/4
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/5
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/6
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/7
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/8
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/9
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!
interface FastEthernet0/10
switchport mode access
spanning-tree portfast
spanning-tree bpduguard enable
!

!
interface GigabitEthernet0/2
switchport trunk encapsulation dot1q
switchport mode trunk
channel-group 1 mode desirable non-silent
!
interface GigabitEthernet0/3
!
interface GigabitEthernet0/4
!
interface Vlan1
ip address 10.5.11.13 255.255.248.0
no ip redirects
no ip proxy-arp
no ip route-cache
no ip mroute-cache
!
ip default-gateway 10.5.15.1
ip classless
no ip http server
!
!
snmp-server community NSGmgt RW
snmp-server community public RO
snmp-server location Dammam Call Center
snmp-server contact NSG 85777
snmp-server host 10.21.249.1 NGSmgt
snmp-server host 10.21.249.1 NSGmgt
tacacs-server host 10.21.6.20 key 7 11081A161C1712
tacacs-server host 10.21.6.21 key 7 14161118070133
tacacs-server directed-request
radius-server source-ports 1645-1646
!
control-plane
!
banner login ^CC

Pleas let me know if anyone faced the same issue!!!! and know the solution

Polycom 320. Can I use RJ9 headset and remove handset?

TheFonze73 — Sun, 08 Nov 2009 06:01:23 GMT

I have 12 Polycom 320 phones. I want to use a headset on all phones and remove the handset. I don't want to use an adapter for the headset minijack on the side of the phone but want to plug directly into the handset socket.

Is this possible?
What settings do I need to change?

Many Thanks

Help..any ideas? (Other party hears dialing while having conversation)

chuckles — Sun, 08 Nov 2009 03:04:48 GMT

I experience the same. Likely cause is "false detection of inband DTMF" by your ATA. Read:
http://forum.voxilla.com/linksys-sip...p2t-32847.html

and
http://forum.voxilla.com/linksys-sip...ion-27370.html

Dial Plan Wizard Software - Beta

nick007shadow — Sat, 07 Nov 2009 17:49:31 GMT

Quote:

Originally Posted by doperz (Post 135127)

Have someone a working download link for this software!

It'll be nice ;)))

chan_iax2.c:13514 load_module: Unable to open IAX timing interface: No such file or d

oddsmojo — Sat, 07 Nov 2009 03:17:09 GMT

Before I get flamed for posting something that has been answered, I have gone through 3 hours of reading through, re-installing the ztdummy, fix after fix and I still can't figure this out. Information as follows:

I am setting up a cardless PBX with softphone capabilities. Board is a Foxconn Quadcore with 8Gb Ram, Raid 1 500Gb etc.

Centos 5.3 ( Linux pb1.blah.net 2.6.18-164.6.1.el5PAE #1 SMP Tue Nov 3 16:55:59 EST 2009 i686 i686 i386 GNU/Linux )

To get started. There is no card in the system, it is 100% SIP based. I am new to setting this up, if I am missing something obvious or if this is not possible please advise. I learn fast :)

Configuration information:

Code:

[root@pb1 asterisk]# lsmod | grep zaptel

zaptel                185892  8 ztdummy,xpp,wct1xxp,tor2,pciradio,wcfxo,wctdm24xxp,wctdm

crc_ccitt               6337  1 zaptel

Code:

[root@pb1 asterisk]# cat /proc/interrupts

           CPU0       CPU1       CPU2       CPU3

  0:    4619153      15201      15217      15202    IO-APIC-edge  timer

  1:          2          0          0          0    IO-APIC-edge  i8042

  7:          0          0          0          0    IO-APIC-edge  parport0

  8:    4532478       4995       4979       4993    IO-APIC-edge  rtc

  9:          0          0          1          0   IO-APIC-level  acpi

 12:          4          0          0          0    IO-APIC-edge  i8042

 50:       5531          0      16844          0         PCI-MSI  ahci

 58:         52         50         49    1202295   IO-APIC-level  HDA Intel, eth0

201:          0          0          0          0   IO-APIC-level  uhci_hcd:usb3

209:          0          0          0          0   IO-APIC-level  ehci_hcd:usb1, uhci_hcd:usb8

217:          0          0          0          0   IO-APIC-level  ehci_hcd:usb2, uhci_hcd:usb6

225:          0          0          0          0   IO-APIC-level  uhci_hcd:usb4

233:          0          0          0          0   IO-APIC-level  uhci_hcd:usb5, uhci_hcd:usb7

NMI:          0          0          0          0

LOC:    4664895    4664894    4665003    4665003

ERR:          0

MIS:          0

Code:

[root@pb1 asterisk]# ztcfg -vd

Line 14: loadzone       = us

Line 15: defaultzone    = us





Zaptel Version: 1.4.2.1

Echo Canceller: MG2

Configuration

======================





0 channels configured.

Code:

[root@pb1 asterisk]# ll /dev/zap/

total 0

crw-rw---- 1 asterisk asterisk 196, 254 Nov  6 00:13 channel

crw-rw---- 1 asterisk asterisk 196,   0 Nov  6 00:13 ctl

crw-rw---- 1 asterisk asterisk 196, 255 Nov  6 00:13 pseudo

crw-rw---- 1 asterisk asterisk 196, 253 Nov  6 00:13 timer

crw-rw---- 1 asterisk asterisk 196, 250 Nov  6 00:13 transcode

Everything loaded right? I have forced out all of the devices and loaded only ztdummy but that made no difference ( as it shouldn't since they are modules )

Code:

[root@pb1 asterisk]# cat modules.conf

;

; Asterisk configuration file

;

; Module Loader configuration file

;



[modules]

autoload=yes



noload => pbx_gtkconsole.so

noload => pbx_kdeconsole.so

noload => app_intercom.so

noload => chan_modem.so

noload => chan_modem_aopen.so

noload => chan_modem_bestdata.so

noload => chan_modem_i4l.so

load => res_musiconhold.so

noload => chan_alsa.so

noload => chan_oss.so

noload => chan_skinny.so

noload => pbx_dundi.so

noload => chan_h323.so

noload => chan_zap.so

noload => app_curl.so

noload => codec_dahdi.so

The problem:

Code:

[Nov  6 00:46:22] WARNING[8050]: chan_iax2.c:13514 load_module: Unable to open IAX timing interface: No such file or directory

"No such file or directory"... I'm really not sure how to approach this. All of the guides I have found state to use the ztdummy zaptel drivers which I have done. Here is what I have found while running an strace:

Code:

futex(0x81a7ec4, FUTEX_WAKE_OP_PRIVATE, 1, 1, 0x81a7ec0, {FUTEX_OP_SET, 0, FUTEX_OP_CMP_GT, 1}) = 1

futex(0x8191b88, FUTEX_WAKE_PRIVATE, 1) = 1

  == Registered custom function 'IAXPEER'

gettimeofday({1257492415, 11460}, NULL) = 0

futex(0x81a7ec4, FUTEX_WAKE_OP_PRIVATE, 1, 1, 0x81a7ec0, {FUTEX_OP_SET, 0, FUTEX_OP_CMP_GT, 1}) = 1

futex(0x8191b88, FUTEX_WAKE_PRIVATE, 1  == Registered custom function 'IAXVAR'

) = 1

open("/dev/dahdi/timer", O_RDWR|O_LARGEFILE) = -1 ENOENT (No such file or directory)

open("/dev/dahdi/pseudo", O_RDWR|O_LARGEFILE) = -1 ENOENT (No such file or directory)

gettimeofday({1257492415, 11716}, NULL) = 0

stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=2427, ...}) = 0

stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=2427, ...}) = 0

stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=2427, ...}) = 0

gettid()                                = 7924

futex(0x81a7ec4, FUTEX_WAKE_OP_PRIVATE, 1, 1, 0x81a7ec0, {FUTEX_OP_SET, 0, FUTEX_OP_CMP_GT, 1}) = 1

futex(0x8191b88, FUTEX_WAKE_PRIVATE, 1) = 1

[Nov  6 00:26:55] WARNING[7924]: chan_iax2.c:13514 load_module: Unable to open IAX timing interface: No such file or directory

gettimeofday({1257492415, 13790}, NULL) = 0

gettimeofday({1257492415, 13984}, NULL) = 0

futex(0x81a7ec4, FUTEX_WAKE_OP_PRIVATE, 1, 1, 0x81a7ec0, {FUTEX_OP_SET, 0, FUTEX_OP_CMP_GT, 1}) = 1

futex(0x8191b88, FUTEX_WAKE_PRIVATE, 1  == Registered application 'IAX2Provision'

Notice the:
open("/dev/dahdi/timer", O_RDWR|O_LARGEFILE) = -1 ENOENT (No such file or directory)
open("/dev/dahdi/pseudo", O_RDWR|O_LARGEFILE) = -1 ENOENT (No such file or directory)

Maybe there is a line in a configuration file specifying which device to use for this? To test this out by linking the zap directory in, it fails as well.

ln -s /dev/zap /dev/dahdi

Code:

stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=2427, ...}) = 0

stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=2427, ...}) = 0

gettid()                                = 8004

write(1, " Asterisk Event Logger Started /"..., 59 Asterisk Event Logger Started /var/log/asterisk/event_log

) = 59

sched_yield( Asterisk Event Logger Started /var/log/asterisk/event_log

[Nov  6 00:38:55] ERROR[8004]: asterisk.c:3385 main: You have DAHDI built and drivers loaded, but the DAHDI timer test failed to set DAHDI_TIMERCONFIG to 160.

)                           = 0

exit_group(1)                           = ?

And it exists immediately.

I hope this was not too much information to digest, I wanted to include everything that I might have been asked for. If I'm clueless and am missing something please let me know, I'm at a loss.

I hope someone can help!

Cheers

- Robert

FS: Brand New BlackBerry Bold 9700 & HTC Touch Pro2 @ Affrodable Prices.

JSOffers — Sat, 07 Nov 2009 02:49:10 GMT

BlackBerry Bold 9700
PRICE: $270

Technical Details.....

Dimensions: 109 x 60 x 14.1mm
Weight: 120g
Battery: Talk Time: 6 hrs Capacity: 1500 mAh
Display: 480 x 360 pixels/2.44"

Network:
2G: 850/900/1800/1900 (Quad-Band)
3G: 800/850/1900/2100 (Quad-Band)
HSDPA (3.6Mbps)

Camera:
3.2 mega-pixels (auto-focus)
2 x Digital Zoom
LED Flash
Image Stabilization -
Video:
30fps Video Recording (480 x 352)
Supported formats: MP4, 3GP, H.264, H.263, DivX, XviD, WMV & Flash
Video Streaming
Music:
Supported formats: MP3, AAC, AAC+, eAAC+ & WMA

Ringtones:
Monophonic
Polyphonic
MP3
WAV

Messaging:
SMS
MMS (with video)
E-mail (POP3, SMTP, IMAP4, BlackBerry, MS Exchange)
Instant Messaging (MSN)

Memory:
Phone Book (unlimited)
Dialled Calls
Missed Calls
Received Calls
256MB (internal)
microSDHC (external)

Call Features:
Hands Free
Caller ID

Connectivity:
microUSB
3.5mm Audio Jack
Bluetooth (2.1)
Wi-Fi (IEEE 802.11g)
Navigation:AGPS

Security:
Device Lock
Remote Lock
Device and memory card encryption
Mobile VPN support.

Features:
Optical Track Pad
Office Document Viewer (Word, Excel & PowerPoint)
PDF Document Viewer

For Bookings and More Inquires on our Sales and services Visit...

Company Name: BEST SOURCE OFFER LIMITED
Contact Person: Josh Richards
Contact Mail: bestsourceofferltd@live.com
Web Page: www.bestsourceofferltd.page.tl

Pre-purchase question for the HT502

neftv — Sat, 07 Nov 2009 01:51:59 GMT

I was wondering if its possible to implement something that the SPA1001 can do.
I want to have two Voip providers. Provider A for 911 and incoming calls and local outgoing calls. Provider b just use for outgoing international calling. Basically, I want to be able to do this since I have one phone say on Port one of the HT502.

VoIP on BlackBerry

edongp — Fri, 06 Nov 2009 21:16:30 GMT

Recent release of Vonage Mobile® application on the Iphone and Blackberry has caused a lot of excitement among bloggers, Iphone and Blackberry users and the news media. What is little known to everybody is that with the installation of a small hardware on the Vonage World® line, Vonage Mobile® becomes unnecessary.Everybody was waiting for a Vonage World® version that would work on their cell phones. What we got was just a cheaper calling card that works only on an expensive smart phone - the Iphone or the Blackberry. Using an Iphone, we have a choice of operating it on a 3g network, WiFi or regular cellular network. With the Blackberry, we can only use the regular cellular network. When we run it on a 3g network, we incur an additional $20 monthly fee. When we run it on a regular cellular network we pay for expensive air time and long distance charges. When we choose to go with WiFi we have to seek out hotspots, which are not readily available. It is also inconvenient that we must anticipate and prepay for these services. There is no need for any of these if you already have a Vonage World® account. You can extend your Vonage World® line to any cell phone now. Here’s how:

You subscribe to a cellular plan which has either free incoming calls or free calls to and from favorite numbers. You designate your Vonage® number as a favorite number. This gives you free unlimited air time calling to and from that number. Even as heavy cell phone users, you can sign up for the minimum $49 per month plan and slash your monthly bill by half and still make free unlimited air time and long distance calls.

To illustrate how Vonage World® can be accessed manually by any cell phone user using the regular cellular network, let us say you have somebody (your wife) at home where your Vonage World line is installed. You call your Vonage® line (or have your wife call you from the Vonage® line), your wife answers and you tell her to transfer the call to another number which could be domestic (all USA and Canadian numbers located anywhere in the world are transferable) or international (some countries like India, China, Malaysia are transferable). Your wife will tap the hook switch (flash), put you on hold and get another dial tone. She then dials “#90” (to tell Vonage® it is a transfer call), the country code, city code and the telephone number. Vonage® will automatically transfer the call to connect you directly to the destination number. Vonage® will send your wife a stutter dial tone and a long dial tone to signal her to hang up and free up the Vonage® line. Your son (friend, employee, etc.), using their own cell phone, can repeat the process and be connected to their destination number at the same time you are using the system. Of course it is not practical for your wife to do all these things for you even if you are saving a bundle. What you can do is install a telecommunication switch such as the DialMate CM1003 (see www.woodtel.com) on your Vonage® line and have it perform all these functions for you. With this combination you can place the cheapest domestic and international calls from your cell phone. As an added bonus, the people you call in those 60 flat-rate-calling-countries can also use your Vonage World® line to call those 60 countries for free and most importantly, using just an ordinary cell phone or landline, without a broadband connection. Replacing your sim card with a local one when traveling to those flat-rate countries that include free cell phone calling gives you the same benefit.

® Vonage Holdings Corporation Registered Trademark

VoipSwitch on Rent / Sale by Solution4VOIP.com

RF.com — Fri, 06 Nov 2009 10:11:06 GMT

Dear Friends:

Want to setup VoIP company, a business under your own brand name? We have complete solution to launche VoIP (Voice Over Internet Protocol) company. All support comes included.
Features: PC2Phone, Device2Phone, Calling Card, Callback,sms callback solution, Ani Callback solution, DID callback solution, Cli Callback, Pin Callback, Wholesale Termination, Online Billing, Unlimited Resellers Creating, online shop, invoice generator, paypal integrated online shop, pin recharge modules, H323 and SIP.
No ties to us, deal with the wholesalers you want and buy VoIP minutes from any one in the market without any expensive equipment.

System

Single platform solution
Rapid Service Rollout
Pre-configured templates
Scalability - phased investment
Quick, flexible reporting
Open database architecture allowing for extended reporting functions and use of external scripts

Switching

Carrier-grade softswitch
Gatekeeper, registrar and proxy functionality
SIP and H323 protocol support
Protocols transparent conversion
Network protection capabilities
Advanced routing
Single entry point into VoIP infrastructure
Flexible proxy methods

Billing

Various authentication methods
Prepaid/postpaid accounts
Robust engine
Fully integration with the softswitch
Cooperation with MySQL or MSSQL

Management interface

Real-time traffic and system's performance monitor
Windows graphical interface
Web-based interface

Web interface for endusers

Customizable customer self care web interface
Admin section with embedded html editors
Ready to use design templates
CDRs, payments, invoices, address book, user's profile and others/
Sending SMS, web callback

IP IVR

Customizable IVR scenarios (XML)
Ready to use templates
Caller ID recognition
One or two stage calling procedures
Support for DIDs as access numbers

Softphone

Based on SIP protocol
Proprietary VOIP tunnel technology - making/receieving calls even behind voip blockades
Fully integration with the web interface
Quick access to voicemail
Sending SMS

Callback

Authentication by callerID or PIN
Various methods of realizing callback
Support for DIDs as callback service numbers
SMS and Web triggered callback
Embedded IVR system

Resellers system (VSR)

Multilevel structure
Web-based comprehensive interface
Support for multi-currency
Customizable web interface for end-users
Integration with the E-Shop

Callshop

Real-time monitoring and billing
Booths visualisation
Standalone and web based versions invoicing

E-Shop

Various payments processors
Credit cards, paypal, moneybookers
Automatic sign-up procedure
Recharge, also by vouchers/PINs
Basket for selling products online

SMS module

Support for http methods
Extended routing plan with support for multiple SMS providers
Integrated with the softphone and the end-users web interface
Support for wholesale SMS traffic

Capacity

Average result 1000 conncurent calls (full proxy) on one server (Dual Xeon CPU 2GB RAM*). For higher traffic recommended is cluster configuration with two or three servers with voipswitch software running on seperate servers and connected to one, shared SQL database installed on a dedicated server. Web services running on a backup SQL server set in replication mode so that the load from web requests (for example browsing CDRs, statistics) does not affect the performance of the main SQL server.

* tested and certified for SIP interoperatibility with Level3 (US based tier one carrier)

Scalability
Phased implementation of new services by enabling additional modules.

Operating system
Windows 2000, XP pro, Windows 2003 (both standard and web editions)

Hardware requirements
Recommended are servers based on Intel Xeon processors

dual core or quad core, in configuration of single or dual CPU,
RAM minimum 1GB,
HDD 100 GB or more

-------- Valid Link to check Before buy -----------------

VSC Panel for Administrator :
http://www.voipswitchdemo.com/vsc/admin/index.aspx

VSR or Reseller Panel :
http://www.solution4voip.com/vsr

Portal for End User :
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VSPortal ( Flashed Based ) For End User :
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WTP ( WholeSale Trading Platform ) :
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VSMedia ( IVR Customization + Encoding ) :
http://www.solution4voip.com/vsmedia

Callshop Custom WebBased :
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-------- Valid Link to check Before buy -----------------

------- VoipSwitch Rental Plan ------------------

30 concurrent calls – VSP Server with 1 GB RAM, 80 GB HDD
( 2 Mbps dedicated bandwidth, Price = 150 USD/month )

50 concurrent calls – Pentium IV 2.4 GHz 1 GB RAM, 80 GB HDD
( 3 Mbps dedicated bandwidth, Price = 250 USD/month )

120 concurrent calls – Pentium IV 3.0 GHz 2 GB RAM 180 GB HDD
( 5 mbps dedicated bandwidth price =350 USD/month)

200 concurrent calls – Core 2 Duo 3.0 Ghz, 1GB DDR RAM 180 GB HDD
( 10 mbps dedicated bandwidth Price = 450USD /month )

300 concurrent calls – Xion 3.0 Ghz, 2GB RAM 180 GB HDD
( 13 mbps dedicated bandwidth Price = 550 USD /month )

500 concurrent calls – Quad Core Prossesor , 2GB RAM, 180 GB HDD
( 16 mbps dedicated bandwidth Price = 750 USD /month )

1000 concurrent calls – Dual Qaud Core 3.0 GHz 4GB RAM 180 GB HDD
( 32 mbps dedicated bandwidth Price = 900 USD /month )

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More...

I need someone fix my dial plan (payed job)

Krayt3ch — Fri, 06 Nov 2009 02:04:56 GMT

Or you could post more info and we could try and teach you how to fix the issue your self

You can give a man a fish, and he’ll eat for a day. You can teach a man to fish and he’ll eat for a lifetime.

EDIT: that is unless you dont like fish, in which case we can try chickens (although they are a little bit harder to catch IMO)

SPA9000 and SNOM phones: Interoperability problem?

daneoz — Thu, 05 Nov 2009 21:36:09 GMT

No problem is trivial if you don't know the answer and it's VERY FRUSTRATING.
I've got the same problem with SNOM 300's but my problem is I can't get them registered on the SPA9000 so they consequently do not have number assignments for incoming. Have you got yours registered?
I do have a problem with intermittant one way audio when dialing out to analogues on SPA9000. No TX, RX okay. Same type of problems? I've tried G711 and G729.
Any problems sound familiar? If so can you help me?
Thanks, Dan

Can admin look into Certificate wizard?

rajil.s — Thu, 05 Nov 2009 17:19:24 GMT

I couldnt find genmc from any reliable source (Linksys/Sipura/Cisco). However, i did find a tool at http://www.megajournal.ru/journal/us....98.tar.gz.mp3 , it did generate keys for the Sipura devices. However, the generated keys are not probably correct since the sipura dont talk securely to each other.

I am hoping the mods would fix up the tool on voxilla, as i have had success in the past using certificates generated from it.