Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. Internet telephony refers to communications services — voice, fax, SMS, and/or voice-messaging applications — that ...
1
vote
1answer
9 views
Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk
We have Trixbox CE 2.8 and Asterisk 1.6.
I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls.
I also want to remove the +1 for long distance so ...
0
votes
0answers
6 views
How to pass/process extension numbers while calling Freeswitch from outside SIP gateway?
Suppose I have Freeswitch, which has SIP gateway configured. So, the local users of this Freeswitch can call outside via
<extension name="russia">
<condition field="destination_number" ...
0
votes
1answer
16 views
3com 3101 or 3102 VOIP phones will not get DHCP/connect to network unless in specific location?
I am trying to figure out why at our company the 3com 3101 or 3102 phones we use, when NEW out of box will not jump on the network and update, unless they are at a specific physical location.
For ...
1
vote
2answers
27 views
FreePBX How to play an announcement for misdialled calls
Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately.
Some of our users who ...
0
votes
1answer
22 views
Linksys SPA901 won't react to ****
Uh… I don't know if this is correct stack exchange to ask, but it's the best one I've found.
I have Linksys SPA901. And when I enter **** (to enter configuration mode) I hear nothing at all. Also I ...
1
vote
0answers
9 views
Why adding capture filters breaks the traffic dump in wireshark / windump?
I have a strange issue while trying to capture RTP (UDP) traffic.
I have a phone using IP 192.168.9.4 and a Windows 2003 PC connected to the same switch (actually to the monitor port of the switch - ...
-1
votes
1answer
39 views
Is a PBX responsible for routing calls, or is that handled by the provider?
I am setting up a PBX that external users will connect to and I was wondering how important PBX placement was.
Are calls routed through the PBX itself, or do they go directly through the provider ...
-1
votes
0answers
43 views
Asterisk E911 Service in UK
I see that Asterisk supports the E911 service where by you assign an address to a number, then the number is sent down a PRI for example, when dialing an emergency number (like 911,) so the emergency ...
0
votes
0answers
24 views
Automatically log into SIP phone when user logs into Windows
We run an asterisk-based PBX, with FreePBX front end, configured in 'device and user mode' to support hot desking users. That is, a user arrives and sits at any desk, keys *11 into the phone (Cisco ...
0
votes
1answer
54 views
How to associate a softphone with a phone url
I'm trying to get click to call to work using Qutecom for a softphone and a Greesemonkey script to turn phone numbers into clickable urls. Whenever I click on the phone number link I get an error ...
0
votes
1answer
65 views
Click to call with a soft phone
I'm trying to set up click to call with a soft phone. I'm using QuteCom for the soft phone, vTiger for the CRM and Asterisk, through Elastix for the PBX.
I was wondering if someone could point me in ...
2
votes
2answers
111 views
Setting up a SIP NAT proxy
We currently run a VoIP server using an upstream providers SIP proxy for our clients who are behind NAT. We now have the problem that we re ending the relationship with the upstream provider, and will ...
1
vote
1answer
152 views
GUI for Asterisk 1.8.8 on Debian 6
Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
0
votes
2answers
210 views
Asterisk + SIP 404 not found
I want to make a small Asterisk server in my house. I installed asterisk on my Ubuntu
and I use 2 computers, in order to connect to one another. when I connect I can see in Wireshark that registrar ...
1
vote
0answers
45 views
Asterisk 1.8, Dedicated SDSL but SIP Provider Inbound Channel heavily delayed?
We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of ...
0
votes
2answers
62 views
Cisco Phone (7961) won't find voice vlan
We have a cisco phone system setup successfully.
I am setting up a bunch of phones, of which, one keeps getting an data vlan ip address from the DHCP server.
Any suggestions?
I have factory reset ...
6
votes
4answers
253 views
Is it possible or advisable to virtualize a PBX system? How would one go about this?
I'm totally new to the world of VoIP and we are looking to move from our current provider to a solution we host ourselves, mainly because the current service is so unreliable. Unfortunately I know ...
0
votes
0answers
36 views
Why doesn't Asterisk even try to reach any IAX peer?
Here is the whole /etc/iax.conf of my otherwise default Asterisk installation on Debian Squeeze :
[general]
disallow=all
allow=alaw
allow=ilbc
allow=gsm
jitterbuffer=yes
autokill=yes
register => ...
1
vote
1answer
99 views
Dell equivalent of Cisco Span
We are in the process of upgrading out network to Gigabit to the desktop.
Currently we are using this equipment...
CUCM (Cisco Unified Communication Manger 8.0.3.20000-2)
Cisco 2821 Integrated ...
6
votes
2answers
145 views
Can you please explain the difference between PBX and soft switch in less technical terms?
Asterisk is an open source PBX and FreeSWITCH is an open source soft switch.
Source: http://www.freeswitch.org/node/117
Can you please explain the difference between PBX and soft switch in less ...
0
votes
1answer
113 views
Are there parallels to Asterisk AMI and Asterisk AGI in FreeSWITCH?
Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also ...
1
vote
2answers
78 views
Separate VoIP telephony and computer network - or not?
We have a PBX server (also connected to the internet) and all VoIP devices currently (early stages of deployment) set up on a network that is phsyically separate from the main computer network.
As ...
1
vote
1answer
42 views
Our VOIP system receives a lot of calls from an “uknown” caller and subsequently crashes
In our office, we have a voip server(sipXecs) that is connected to a few phones and ATA boxes.
Recently, our VOIP system has been giving us problems. On first boot, it's fine, but then after a while, ...
3
votes
4answers
80 views
Using the Built in Switch in IP phones
I'm working on setting up a small business with IP Phones to connect to the Asterisk Server. Now, for simplicity we just ran 2 CAT5e lines to each room, and 1 was hooked to a phone jack, before my ...
0
votes
1answer
35 views
Make VOIP-phone Cisco SPA504G to receive only 1 call
I don't want the phone to receive more that 1 call. So if an operator have one ongoing call, a new incoming call will received busy signal. As far as I understand, there is no option to set this ...
1
vote
2answers
68 views
Advantages and disadvantages to using a VPS (or any remote server) to host a VOIP/PBX system
My company currently runs a Trixbox Pro server locally. We're getting sick of the licensing fees and terrible community support and are considering migrating to something like FreePBX. I see that ...
0
votes
0answers
48 views
Asterisk PBX system for VoIP
we are currently moving into having our VoIP on a PBX box on our system.
What I would like to do is setup a server on the AWS cloud that will sync with the one here locally and load balance to ...
0
votes
0answers
52 views
Off-site voip phone locking up, and I suspect firewall issues at the remote site
My Aastra 6757i off-site voip phone locks up periodically, and I want to rule out a networking problem. Most of the time calls work fine, both placed and received calls. But sometimes the voip call ...
-2
votes
1answer
68 views
Asterisk - which protocol/technology is using? [closed]
Asterisk installed on Linux server by other admin.
I am trying to find out which communication protocol/technology is currently using on Asterisk? Eg: isdn, voip, acd, etc. I dont know which one is ...
1
vote
1answer
73 views
Using Asterisk and FreePBX how can I map extensions to outbound routes
I have a Trixbox server (Asterisk and FreePBX) that has multiple tenants on it. I need these tenants calls to go out via different outbound routes in order to split the bills at the SIP trunk provider ...
0
votes
1answer
30 views
AsteriskNow Migration / Shared Extension Space
I am testing the possibility of migrating from an old Avaya phone system to AsteriskNow. The migration would cover several hundred phones--but spread out over several years. (Management wants to ...
2
votes
4answers
139 views
Intercom over IP?
This is an odd request, but I'm hoping some of you have had to field it before:
We have an office in New York and remote employees in Oregon and Florida. Communication is extremely important to us, ...
0
votes
0answers
17 views
Reliable applicance for routing IT emergency calls (SIP and ISDN)
We have a fairly big IT installation and our IT staff needs to be reachable 24/7. At the moment we have the following setup for "emergency" calls to our IT staff on our main Asterisk box:
An ...
1
vote
2answers
49 views
How can I filter packets from a port monitor?
I have some data going from Point A to Point B. I have a SPAN monitor set up to a monitoring device C.
To recreate some real world scenarios, I want to filter out all traffic which is a certain type ...
0
votes
1answer
32 views
How can I disable the audible MWI on Polycom phones?
By default, when a Polycom VoIP phone has a voicemail message waiting, it both lights the MWI light, and plays a sound every several minutes. I've never met a more self-important feature in a phone, ...
1
vote
2answers
102 views
Asterisk and sip station codecs
we have a small instance running on ec2 of
voxilla/FreePBX-Asterisk-1.6.0.5-i386.manifest
which has been updated as far as i can see
We signed up for sipstation (because it removes half the ...
0
votes
1answer
14 views
Does ENUM make a measurable difference in percentage of calls routed via IP vs PSTN?
I'm considering configuring this, but don't want to waste time if there isn't a significant cost saving to be made. Any advice would also be much appreciated.
Thanks,
Chris
2
votes
1answer
79 views
how is the voip network working when server is outside nat and clients are inside nat?
I want to install an asterisk on a server with a public ip. The sip clients are behind different NATs. How is the signaling and the rtp packets (the voice) working in this case? Will rtp be peer to ...
0
votes
2answers
83 views
how to associate an ip with a name?
I have installed an asterisk on my private network(so it s behind a router ). Is there anu way to set the asterisk on a public ip though it's installed in a server with private ip? As Ive read it ...
0
votes
0answers
72 views
Is it possible to split Asterisk SIP signaling and voice traffic?
How do I configure Asterisk to send SIP signaling to one IP address and the RTSP stream, voice traffic, to another address?
1
vote
0answers
115 views
Site B FXO configuration for incoming calls
I am trying to configure the FXO port (voice-port 4) on a Cisco SRST881. Currently, there is one working remote extension here (ext 50) and all calls originate from our main office.
My goal is to ...
0
votes
2answers
275 views
Asterisk - inbound calls from SIP DID trunk “rejected because extension not found”
I have a DID with DIDforSale that's pointed at my Asterisk server. When I call it from my landline phone I get the AT&T disconnected line recording. The Asterisk CLI shows the error message:
[Oct ...
1
vote
2answers
127 views
Asterisk DID incoming rules =s
So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
0
votes
1answer
63 views
VOIP network and Computer Network together
I have two networks, my main data network which is a domain controller and my VOIP network for phones. Both networks have completely seperate hardware (routers,switchs, e.c.t) I woul like to have ...
1
vote
1answer
41 views
Linux audio capture at a rate of 20ms is choppy
I am currently working on an embed linux video server. When I deployed the application to the boards we are planning on using for production I noticed that the audio was noisy.
I discovered that the ...
3
votes
2answers
119 views
Extremely Poor VOIP Audio Playback Performance During Live Phone Calls with Remote Desktop Services
The setup:
Plantronics USB CS50 Headset > USB Port of "Thin Client" (Running Win Pro) > (Wireless) RDP to Remote Desktop Server > Eyebeam Softphone Client > VOIP Server > Caller (cell phone or ...
1
vote
1answer
82 views
I need some advice to come up with a network plan, or sort of anyway
At my workplace we have been allotted a few IP addresses by the ISP. I think currently one IP is accessible from the outside (by accessible I mean in a browser). Typing this IP into a browser gets you ...
0
votes
1answer
22 views
How do I change outside access number from 8 to 9 in Cisco UCM?
I want to change my outside access number from 8 to 9 in Cisco Unified Communications Manager. I've copied all of my 9.x route patterns to create a duplicate set of 8.x route patterns.
When I try to ...
0
votes
0answers
12 views
searching for unified communication platform
I'm searching unified communication platform solutions that have features VOIP, instant messaging, file sharing, something that works like fring (video chat are optional, ...
0
votes
2answers
37 views
Calling out from Asterisk Server
I have an Asterisk server connected with a Legacy PBX. This setup is connected to another Asterisk Server (In another country) over the Internet with an IAX2 Trunk. Inter Office Extensions are ...
