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 ...
3
votes
4answers
106 views
What does “non-IP” originated traffic mean in a TOS?
I am looking at VoIP providers and many of them have a part of their TOS that says they don't allow "non-IP originated traffic."
Excerpt from TOS:
You shall only send to [DELETED COMPANY NAME] ...
0
votes
0answers
32 views
Choosing best SIP server [closed]
On what factors my choice for SIP server should depend. There are so many open source as well propriety servers for WINDOWS/linux machines in market. Can someone point me in right direction.
Thanks
2
votes
0answers
28 views
sip endpoint receives direct sip calls but doesn't receive redirects from the registrar
I have an SIP endpoint, which I wrote, that receives sip invites, processes them, replies, and sets up the rtp session just fine when contacted directly.
i.e.
sip:user@[actual ip address of ...
2
votes
1answer
34 views
How to set up QoS VoIP priority on linux
I am configuring a small network and my idea is to use a linux server as a proxy it will be serving local services to the clients and it will have also a squid server as a caching server for speeding ...
1
vote
1answer
65 views
HP Procurve to Cisco - QOS
I have a HP Procurve with the following:
vlan 4
name "Phones"
untagged 20
ip address 10.50.60.1 255.255.252.0
qos dscp 101110
qos device-priority 10.50.100.10 dscp 101110
qos ...
2
votes
3answers
62 views
How to add more incoming/outgoing lines capability to Asterisk?
We have 4 people call us from around the country at once and it causes our phones to be busy.
We want to be able to have at least 8 people call us at once without anyone getting the busy signal. ...
0
votes
1answer
53 views
Get Asterisk/Freepbx to ignore digits that were unnecessarily dialed (area code for local calls)
This is the second part of this question, worded better.
Remove local area code on outbound calls, and remove "dial 1" for long distance in Trixbox/Asterisk
I have removed the NEED to ...
1
vote
1answer
30 views
How to send Voice IP traffic over a different DSL router
We have the following equipment for our internet and voip access
2 x netgear ADSL routers
1 x DHCP / DNS server (Windows 2008 server)
Numerous PCs
VOIP Phones
I want to configure our network so ...
0
votes
0answers
38 views
Internal VOIP to external PSTN
I need to install 5 telephones in our new offices, however, we have very limited bandwidth so I don't think a VOIP provider will be a realistic option. I have had two standard land lines installed for ...
1
vote
0answers
26 views
FAILED status in Elastix CDR reports
What are all the possible reasons on Elastix to log an inbound call with 'FAILED' status?
I'm generating a CDR report and there's a bunch of calls (around 30%) with this status. Additional values of ...
1
vote
1answer
74 views
How to add a user who is out of state as if they were just another extension AT our office? [closed]
We want to add a person who is in another state to our phone system as though they were simply in another cubicle here at the office. So, they can dial "302" and get the office assistant's extension ...
1
vote
1answer
35 views
NAT setup for an SIP Registrar/Proxy
I am currently trying to get the following scenario to work.
Warning: I am a software engineer--not a network admin.
I have various SIP endpoints (sip based video servers) on a LAN. On a Windows ...
2
votes
1answer
42 views
Lync and conference phones - do all clients need phone numbers?
Bear with me here - very new when it comes to Lync :)
We have a very simple installation up and running right now, with only Lync software clients installed. I've now been requested to install two ...
0
votes
1answer
32 views
Extension not found error on calling an extension
I have setup a SIP server and connected two polycom soundstation duo devices to it. They both register line with the SIP server fine but when I try to place a call from one device to the other, ...
0
votes
1answer
34 views
Voip, connect to asterisk servers over WAN with A580 base station
I have LAN infrastructure with asterisk and SIP phones(Gigaset A580) in it, I have done port forwarding on my public static ip address to my asterisk server in my LAN network, I also have trunk which ...
0
votes
0answers
28 views
Hosted VoIP Issues
I recently switched my company to a hosted VoIP system and we're having some issues that I'm having a hard time resolving myself.
The phones periodically lose connection with the remote server and ...
0
votes
1answer
112 views
How to configure Transparent IP Address Sharing (TAS) on a Mediatrix 4102 with DGW 2.0 firmware?
I am making the switch to VoIP. I chose voip.ms as my service provider and Mediatrix 4102 as my ATA.
One reason why I chose the Mediatrix over other popular consumer ATAs is that it's supposed to be ...
0
votes
1answer
42 views
Setting up office phone system (small business)
I've been asked, within six months, to deploy a phone system. The exact wording was "a phone on every desk." They want inter-phone transfers, external calling, and even software IP phones for our ...
-1
votes
1answer
40 views
Voice over ip setup for small business [closed]
I have no experience with Voip but if I wanted to use VIOP in a business with around 50 viop phones spread over 3 offices would I need to use some sort of voip Server or do most routers these days ...
1
vote
1answer
53 views
IM and File Transfers Integrated With VoIP
I have an Asterisk box serving an office of people. I'd like to provide instant messaging and file sharing capabilities alongside the voice and video capabilities provided by Asterisk (ala Windows ...
0
votes
0answers
81 views
Migrating a Cisco 7940g from SCCP to SIP firmware
We have a number of Cisco 7940 handsets that have come from various sources that we're reusing on an Asterisk server.
We've managed to convert all of these servers from Call Manager (SCCP) to SIP ...
0
votes
2answers
99 views
Practical difference between CISCO C2960PD-8TT-L and Netgear ProSafe GS108P
So I'm setting up a new department and doing some shopping for our new 4 developer team. I need to get some switches so that each member has at least 2 ethernet ports +1 with POE for an IP phone (1 ...
0
votes
1answer
57 views
Router Not Passing *Some* Requests
Sometime yesterday, my Asterisk box quit talking to my voip provider. I've determined it to be the router, as I have a mirrored router on standby should this one causes problems - which it did.
...
0
votes
0answers
41 views
how can i call my voicemail in bluebox freeswitch
I'm using bluebox to manage freeswitch, how can I call my voicemail, if I Dial my extension it takes my to the part where I need to record a voicemail.
if I try and Dial 4000, *89 or 879 its just ...
0
votes
2answers
75 views
Linux SIP Client for Server/Bot
I need a solution for just receive a call from a SIP Provider and answer the call automatically with a mp3/wav file.
I've already achieved this using a complex and big PBX solution like Freeswitch, ...
0
votes
0answers
39 views
Connection Lost between phone and CallManager
a customer of mine has a series of phones connecting over an IPSEC vpn over ADSL to CallManager 4.2.
The phones are deregistering periodically, and in the phone logs I see errors like this:
NOT ...
1
vote
1answer
88 views
GMAIL ports/protocols
I am trying to configure iptables on Ubuntu. I have the server setup as a transparent proxy using Squid which is working fine. At this time only port 80 is redirected to SQUID via nat iptables. I ...
0
votes
1answer
66 views
Converge Voice and Data networks using Sonicwall
We are looking to converge VOIP and Data traffic onto a single wire so that our client's VOIP phones pass data through to the users computer. We are specing out a new Sonicwall NSA appliance to handle ...
0
votes
0answers
111 views
OpenVPN Configure on DD-WRT router
I have a TL_WR1043ND router with DD-WRT firmware.
I've successfully configured OpenVpn on my router. I've got openVpn from a commercial VPN provider. I VOIP GSM gateway is connected with the eht0 ...
1
vote
6answers
250 views
How to record SIP traffic / calls for future auditing?
We have a VOIP (SIP?) phone system and have to record all calls for specific phones in the company. It may be required at a future date to listen to these calls for auditing. These phones happen to ...
1
vote
1answer
36 views
Is there a way to get an “extension board” for a Polycom IP6000 phone?
I don't see any for sale, but is there a way to get one hooked up to it? Basically, I need 15+ speed dials.
Something like this would be great, but for the Polycom instead of Grandstream:
...
0
votes
2answers
39 views
How do you set up speed dial on Polycom IP6000 phones?
Basically I would like speed dial on this so I can press 1 and it will dial 101 AND I can press 2 and it will dial 555-555-5555. Is there a way to do this?
1
vote
2answers
115 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
50 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
126 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
75 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
55 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
30 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
51 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 ...
0
votes
1answer
57 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
117 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
135 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 ...
3
votes
2answers
218 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
305 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
662 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
113 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
147 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
304 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
95 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
150 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 ...
