Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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Eviron: Asterisk 11 / FreePBX

This seems simple enough, but I must be using the wrong search terms ... because I'm coming up blank!

I have an application that performs certain actions at the start and end of a call, using pre-dial and hangup handlers. That part is working. Now I'd like to perform a different action, in the hangup handler, if the call went to voicemail. Are there any variables/functions/...that could tell me whether a call went to voicemail before it is disconnected and the hangup handler runs?  

ANY help would greatly appreciated!
0
I did data and voice vlans integration. But I have never done the separation of voice and data vlans. My firm wants to purchase new 3750X switches and have those switches handle just voice. Therefore my internal LAN will have two networks, one for the voice and one for the data. I will have a data core switch and a voice core switch. Those core switches will terminate to a firewall.
My questions are:
- Do you still need to deploy QoS on the voice switches?
- I may separate the voice and data. But when it comes to the FW, I still have the bottle neck. Correct?
- The separation between the voice and the data vlans is just for the internal LAN, when the packets get to the MPLS network, I have to relay on the MPLS provider for the QoS . Correct?
- I am just wondering if there are any other advantages to separate the voice and the data, other than making the traffic more predictable.

Thanks
0
Years ago I bought an Obi 110 and dropped my land line, using Obitalk with Google Voice.  I loved Google Voice because it integrated so completely with the computer (because I am at a desktop computer 24/7).  

Where do I go now?
I liked simul-ring (so a call to my land line number automatically rung my cell phone)
I liked the voice transcription of messages.
I liked being notified by email of new voice mails.
I liked the dashboard where I could view incoming and outgoing call history.
I liked the ability to block certain parties.
I liked having "notifications" of Google Voice activity in my Chrome Browser.
I liked the free phone calls.

Where do I go now?  I don't have to continue using Obi -- if there's an alternative that works.
My brother did some research and he found "Anveo" that works with Obi for about $90.00 a year, but it only allows 1000 minutes of phone calls before excess charges kick in.  I looked at it and it seems complicated -- and there is NO HELP, NO USER'S MANUAL -- that I can find.  And googling "terms" that Anveo uses on its pages does not reveal definitions that help me understand what x means and how x should be deployed.  (I.E. I'm told Anveo allows for simu-ring, but I can't find anything that explains how to initiate it. )
0
Dear Experts,
I have Asterisk on Debian 12.0. The installation process is OK and I also verify through command in asterisk that I am successfully registered to localphone.com. I don't know how to configure the extension.conf file. Below is the configuration for one extension 1015 in /etc/asterisk/sip.conf. Can please somebody guide how the format of extension.conf is working. Before I used to direcly call from the IP phone without dialing any extension but unfortunately the previous server is crashed.

[Internal]
type = friend
callerid =  Purchase
secret = J2j526
host = dynamic
canreinvite = no
context = internal
dtmfmode = rfc2833
mailbox = 1015
disallow = all
allow = ulaw
transport = udp


FILE extension.conf

[internal]
exten => 1015,1,Answer()
exten => 1015,2,Dial(SIP/1015,60)
exten => 1015,3,Playback(vm-nobodyavail)
exten => 1015,4,VoiceMail(1015@main)
exten => 1015,5,Hangup()

Below is the error I am getting
[May 15 17:05:14] NOTICE[2161][C-00000000]: chan_sip.c:25704 handle_request_invite: Call from '' (10.10.1.50:5060) to extension '65703884' rejected because extension not found in context 'public'.
0
I have a spa525g that i would like to make work with cisco call manager 9.1    Is this possible?
0
Dear Experts,
My server is totally crashed. I don't know how the setup was done. Please someone can provide me the setup and configuration guide. Also, I don't know what are the configuration files need to edit. I have account at localphone.com.
Regards.
0
We use NEC SIP with a dedicated fiber line for VOIP calls. We are having issue for the VOIP handset which has crackling sound, word breaks up, we can't hear well when our client is talking.

our switch is using Cisco 3750, for the moment, there is no Qos in place. The NEC VOIP handset indicates there are packet lost. Does it due to no Qos in place? How import is it? thanks
0
Looking for some help on the following issue. I have some special Mailboxes in Asterisk which don't have extension like General Mailbox etc.
I was able to set them up just fine but the problem is i don't want to record the greeting over the phone but want to use a professional recorded file. Problem is that the unavail greeting is stored in Blob format in MYSQL DB.  Question

a) how can i download and upload a new file to update the blob field in the db
b) is there a way to configure Asterisk to store all greetings locally in dir instead of Blob on mysql ?

Thanks

Alex
0
I have been troubleshooting an intermittent problem with one-way audio on proxy phones in a remote office.  I put in dedicated Internet access for the little office to rule out inbound circuit saturation.  An I have added ping check of the gateway from the main site to make sure packet loss isn't causing an issue.  The Cisco trouble-shooting guide suggested looking at routing.  But I do mostly a mix of static and HSRP and my routes are very predictable.  So I don't think that's the issue.  I started thinking that perhaps transcoding resources perhaps are drying up when these one way audio calls go down.  Is there a way to see via syslog or other whether transcoding problems have been happening at the time of these problem calls?  Thanks.
0
Hi can you track the usage of media resources in Cisco Unified Communications Manager 7.x?  Is there an OID I could query with SNMP?  I don't see anything in RTMT to show me media resource usage.  Any insight appreciated.  Thanks.
0
I have skypein which means I can receive phone calls

windows7: where are skype voicemails stored because sometimes I can not play newest using skype interface
0
Dear Team,

I have a CISCO router 3945 E running IOS ver sion 15.2T with UC  license activated.

We are planning for an IOS upgrade from 15.2 to 15.4.

My question is, after upgrading the IOS do we need to install the UC license once more or it will remain the same???
Kindly help me with the Licensing part after upgradation.

Thanks & Regards,
LAJAN JALEEL
0
Hi Experts,

I'm trying to get a ring group to call an external number. I included the # at the end of the external number as instructed but it still doesn't work. Only the internal numbers work but not the external.

I only have one outbound routes that has sort of a catch-all with 'X.' as the dialing pattern.

I currently have the following setup:

FreePBX Server Spec:
1. FreePBX v 2.11
2. Asterisk 11.8
3. CentOS 6.5

Can you please help me understand what I'm missing?

Many Thanks,
Ricky
0
Hello all

we currently get calls via a number of sip trunks (voicepulse, voiceflex etc) to our asterisk pbx (1.4.7.1)

at the moment a call comes in via the voicepulse, an is directed to a sip user

i need to change it so as the call comes in, it forwards to a external phone number via the sip trunk - if that makes sense?

what i was thinking was

the call comes in

[voicepulse-in]

exten =>  12345678910,1,GoTo(Internal,10,1)
exten =>  12345678910,n,Hangup()

Open in new window



the processing bit
exten =>  10,1,Set (MOBIL=${EXTEN} )
exten =>  10,n,GoTo (USA_Local,_XXXXXXXXXX,2 )
exten =>  10,n,Hangup()

Open in new window



the dialling bit

[USA_Local]
exten =>  _XXXXXXXXXX,1,SetMusicOnHold(USA)
exten =>  _XXXXXXXXXX,n,Dial(SIP/voicepulse/1${EXTEN}]
exten =>  _XXXXXXXXXX,n,Goto(s-${DIALSTATUS},1)
exten =>  _XXXXXXXXXX,n,Hangup()

Open in new window



would this work?!

thanks
0
Hi
I am working on Catalyst 3750X and C3560X and I need to create VLAN for Voip phones.
Have no problem to configure VLANs but never did it for phones and I am wonder if there are any  other configuration that will apply for phones.

Also if you could share with configuration for QoS what usually you set up to protect voice I will appreciate much. (Should I use ACLs or NBAR2 ?)

I will need to allow this voice VLAN over ASA-ASA IPSec VPN. Any recommendation for that? Do I need do some settings on ASA to protect voice VPN?
0
I have a few satellite offices where I drop ip phones. I really like the zywall usg line but I've had some problems with SIP.

If I enable SIP ALG and SIP transformations the phones work but I've been receiving ghost calls that my provider is telling me are not coming from them. I tried adding a firewall rule to only allow SIP traffic from our SIP server but I don't think I understand enough about how SIP ALG works to create a proper solution within the firewall. The only thing that works to get rid of the ghost calls is to disable SIP ALG and setup port forwarding to send SIP traffic to one of the ip phones. The issue now is that I am limited to sending the signal from the SIP server to only a single phone where I really need to be able to send SIP traffic to all of the phones on the LAN.

Any feedback is appreciated as to how I may be able to accomplish this!

Thanks
0
I have a Google Plus account and see a list entitled "Hangouts".

How do I het configured for an iPhone conversation using Google Hangouts?

Download and app then log in?

Which app? What steps?

Thanks.
0
And can I do this from my iPhone? I assume I'd need wifi for the iPhone conversation to work?
0
Hi

Recently I have got call from ATT fraud mgmt. saying there was some international calls made using our phone lines.

I have a trixbox CE that is behind the router . IAX2 port is forwarded in the nat statement. And explicit allow on IAX2 port from my other office IP addresses. Rest of the world will get deny for IAX2 ports.

SIP is totally blocked. No HTTPS/HTTP Natting.

So when I checked I have found someone was able to create an IAX2 Extension and made calls to those foreign numbers.

So I am thinking how is it possible. I don't have anyone in my office that has access to any PC or device to be able to it locally.

Also all my CDR prior yesterday was gone too.

I am a bit cranky as I know I get hundreds of requests getting denied for port 3389 and IAX2 ports.

How and where to start is a good Q?

Thanks
0
okay, im looking into deploying this across our business to laptops and tablets.

So far i can see i need to use skype manager to maintain the business users.  Fine no problems.

Now i'd like to deploy it via GPO (correctly). The latest version 6.14.32.104.
Also be able to use GPO or registry settings to lock down the large file send and any other bit i may find.

But, i don't see any central location for this anywhere.

ANyone help me?
0
has anyone used a faxfinder with the cisco call manager?  i've read various documentation about having to change the impedence to 600 complex which i've done.  but still not working, faxes go to the general admin maiblox and not to the user.  i believe it's not sending the dtmf tones.    using mgcp vg204 gateway
0
2x similar sites configured.  
Site A (HQ) using a Cisco ASA 5505 firewall and an HP Switch as the GW for all clients.  2x vLans 01 (data) and 20 (voice).
Shoretel Director, E1k, Sg90 and sg90bri here on vlan20 172.16.0.0/24

Site B (remote office) no DC's with Cisco ASA 5505 and HP Switch (with vlan 1 and 20 again).
shoretel sg90 here.

Calls between LANs over VPN are fine.

Our remote users (pickup a 10.255.255.0 IP) who connect into the firewall are having issues getting voice to work between them and users at Site B.

Ideas?
0
I am trying to enable VOIP (NEC SIP port 5060) on an existing site to site VPN in our branch office.

Branch office uses Cisco 1900 router, that connects with main office with a Cisco ASA 5500 FW.

I can manage to login to the VOIP phone and ring other colleagues from the branch and listen to voicemail, but can't hear each other.  How to enable the VOIP traffic on the router and ASA? Should I enable sip, tcp or udp traffic? How? Thanks


here is the extracted config on the Cisco 1900:

crypto map OCA_VPN_BRANCH 10 ipsec-isakmp
 set peer *.*.*.*
 set transform-set ESP-3DES-MD5
 match address 100

ip nat inside source list ACL-NAT interface Dialer1 overload
ip route 0.0.0.0 0.0.0.0 Dialer1
ip route 10.188.0.0 255.255.0.0 *.*.*.*
ip route 192.168.0.0 255.255.0.0 192.168.10.2
!
ip access-list extended ACL-NAT
 deny   ip 192.168.0.0 0.0.255.255 10.188.0.0 0.0.255.255
 permit ip any any
ip access-list extended INTERNET_PROTECT
 permit ip 10.188.0.0 0.0.255.255 192.168.0.0 0.0.255.255
 permit ip host *.*.*.* host *.*.*.*
 --More--          permit udp any eq bootps any eq bootpc
 permit gre any any
 permit icmp any any echo
 permit icmp any any echo-reply
 permit icmp any any traceroute
 permit tcp any any eq 443
 deny   ip any any
!
access-list 10 permit 192.168.0.0 0.0.255.255
access-list 11 permit 192.168.11.246
access-list 23 permit 10.10.10.0 0.0.0.7
access-list 100 permit ip 192.168.0.0 0.0.255.255 10.188.0.0 …
0
HI,

I want to add a SIP trunk between my asterisk and Cisco UCM 8.5. I read on some forums that it'll not work properly. Could you please advice the best procedure to do the same.

Thanks in advance.
0
Hi All,

Hope somebody can help. I'm struggling a little with incoming Translation Rules on a CME. I have attached my current config.

Basically internally we have a range or Extensions:

6201 to 6299 and
7201 to 7299

We have recently had a PRI fitted with a 100 DDI's

405 9300 to 405 9399

I'm not bother about DDI for all users as this is not required. I would however like to be able to assign Outside Numbers to various users for DDI.

For example:

405 9312 should go to 6249
405 9399 should go to Hunt Group with Pilot 6000

How do I achieve this.

Also how do I assign outgoing channels. My attached Config is working for outside calls but which channels does it use? Any available? If possible I would like to assign say 6 Channels from the PRI for outgoing.

Hope somebody could help. I have looked everywhere and nothing seems to fit what I need.

Cheers For Any Help In Advance.
putty.log
0

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).