The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) ...

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1answer
2k views

What would cause SIP traffic to be seen going into a switch but not coming out?

Background I have been struggling to get my SIP phones to register behind a brand new router and switch in our brand new office. Our PBX is hosted offsite. I have worked with our provider to attempt ...
13
votes
2answers
663 views

not able to register sip user on red5server, using red5phone

I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip And I got -- Registering ...
7
votes
4answers
1k views

Skype-SIP gateway

Does anyone know if you can setup a Skype-to-SIP gateway on your network? I tried Uplink, which works wonders, but only on Windows and with Skype on the same machine... It would be awesome if you ...
6
votes
4answers
2k 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 ...
6
votes
8answers
44k views

How can I check whether port 5060 is open in centos?

How can I check whether port 5060 is open in centos? How can I test if my linux has real a real IP address and I set no iptables blocking rules or is there any tools which I can run in my linux so my ...
5
votes
2answers
1k views

Ways to monitor SIP termination on an asterisk server

I have a nagios setup which ensures that SIP is responsive on my Asterisk server, that's straight forward. My question is, what kind of possibilities are there that the Asterisk server can actually ...
4
votes
5answers
3k views

Command line SIP dialer

Is there a simple command line SIP dialer for unix which can connect to SIP server, make a call and play some media file (wav/mp3)? In ideal I would look like this: sip-dailer +1xxxxxxxxxx ...
4
votes
3answers
3k views

Unregister SIP UAC message

I've looked so much on the internet, but I could not find a any SIP Unregister example, and when I search RFC 3261,3665 the word does not even appear, perhaps I'm searching for the wrong phrase. I ...
4
votes
8answers
9k views

How can I simulate a modem over a VOIP connection?

I have a hardware device I need a server to dial into at regular intervals. The problem is I no longer have a POTS line or a modem in any of development computers, and all my production servers are ...
3
votes
6answers
1k views

What bandwidth would be required for 55 VOIP Lines And what type of Internet connection

In a call centre environment, what bandwidth would be required for 55 lines? This would be on SIP protocol, would we be best to use G729 codec as i know that our sip provider supports this? What type ...
3
votes
4answers
837 views

Asterisk Intrusion Prevention

Let me start of by saying that I'm a noob, and what I've figured out so far has only been by stumbling my way through it. I have Googled around and the solution may be out there already, but it was ...
3
votes
2answers
2k views

How can I make Asterisk keep track of dynamic SIP agent statuses?

I am setting up a new server using Asterisk 1.8.11-certified4. In testing, we're seeing that agents dynamically logged into the queue will receive a second queue call as a call-waiting when ...
3
votes
1answer
140 views

Does anyone know a good Website comparing all the PBX/IP telephone solutions?

I'm looking for a good website dressing a comparative analysis of the most popular telephony solutions resellers (Cisco, Avaya, Siemens, Microsoft, etc...). I already found a very nice work comparing ...
3
votes
3answers
245 views

Cheap VoIP Phone Recommendation [closed]

I'm looking for an inexpensive corded VoIP phone for my home office that supports SIP. I've been using an old Cisco 7905G with SIP firmware but I'm concerned about security considering that it's ...
3
votes
2answers
6k views

Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's ...
3
votes
1answer
5k views

Asterisk SIP/2.0 401 Unauthorized

I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well. The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of ...
3
votes
2answers
2k 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 ...
3
votes
4answers
10k views

How to stop registration attempts on Asterisk

The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:912@xx.xx.xx.xx>' failed for '37.75.210.177' - ...
2
votes
2answers
351 views

VOIP Codec Preference

Do any of you have a preferred codec for VOIP traffic? I guess this is another case where the answer varies depending on use case, equipment, topology, etc... I'm trying to find an optimal codec to ...
2
votes
1answer
214 views

Router gets disconnected once I terminate my SIP application

Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller ...
2
votes
2answers
112 views

SIP calls voice tremble when switch is busy

I have a small flat LAN based on 1Gbit switches from 3Com. My central switch in server rack is connected to a couple of virtualization servers VMware ESXi and PBX Alcatel. Not long ago I started to ...
2
votes
1answer
287 views

Does the Mitel 3300 support direct SIP Trunking?

We have a Mitel 3300 PBX and we're looking to replace our PRI and BRI lines with SIP Trunking. I've seen that the 3300 supports SIP, but does it support direct SIP from a prospective ISP? Thanks
2
votes
1answer
214 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 ...
2
votes
2answers
9k views

Asterisk SIP digest authentication username mismatch

I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue ...
2
votes
1answer
895 views

asterisk/freeswitch in nat/no-nat setup

my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...
2
votes
1answer
884 views

Adding more than one IP to sip.conf tcpbindaddr and udpbindaddr (Asterisk 1.8)

The defaults in the sample sip.conf under Asterisk1.8 contain udpbindaddr=0.0.0.0 and tcpbindaddr=0.0.0.0. I want to bind the incoming [foo] extension to udp:192.168.1.1/255.255.255.0, the outgoing ...
2
votes
1answer
747 views

Lync 2010 SIP Trunk Providers

I'm playing around with a small Lync implementation in my home office and would like to try replacing my 3cx server with it. Currently I use callcentric as my VoIP provider, but based on another ...
2
votes
1answer
1k views

FreePBX: Basic config?

I've just successfully set up an Asterisk+FreePBX install on an Amazon EC2 instance per this guide: http://voxilla.com/2009/10/15/voxillas-freepbx-in-a-cloud-step-by-step-1457 (I've also assigned it ...
2
votes
2answers
912 views

SIP and NAT routers?

SIP was not built with NAT routers in mind, and I'd like to get to the bottom of this issue to check what needs to be done on all devices so it works with NAT routers, and understand in what context ...
2
votes
4answers
175 views

Obtaining local telephone numbers in the U.S. and worldwide over VoIP/SIP

We have a need to obtain local numbers throughout the U.S. and worldwide terminated at our US headquarters. It looks like this service is only available when using Voice over IP due to regulations on ...
2
votes
0answers
93 views

RTP analysis - Discerning ptime (packetization time) for a given VoIP packet capture

I would like some help on the subject of an automated way of discerning the average packetization time (ptime) of a VoIP call's packet capture. The reason I am not depending on the value in the SDP ...
2
votes
0answers
54 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
2
votes
1answer
593 views

MS Lync on Terminal Server

Is there a way that I can install (hosted) Lync on our terminal server and when a user logs on, it automatically inserts their email address and the correct manual SIP settings so I don't have to log ...
2
votes
1answer
477 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
2
votes
0answers
276 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
0answers
1k views

sending faxes over sip with t38modem and hylafax

I would like to send (and not to receive) faxes. AFAIK, t38modem + hylafax should work fine. After configuring hylafax to look for ttxT38-1,ttxT38-2,ttxT38-3, I'm running the following: # t38modem ...
2
votes
0answers
411 views

How can I record a call transfer with freeSwitch?

I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, ...
1
vote
5answers
833 views

Is there a stable cross-platform softphone? Or a Linux-only one?

We use an Asterisk based solution as our PBX, and although we've found hardphones work flawlessly, we have yet to find a reliable softphone. The basic requirement is that it allows multiple calls and ...
1
vote
6answers
3k 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
56 views

Is SIP protocol routeable

I have a sip-able PBX. My Sipdroid phone works on local network. From outside I route the traffic thru a firewall. Even all rules are ok I cannot connect to the PBX with sip when traffic is routed. ...
1
vote
1answer
111 views

SIP me@domain.com vs (123) 456-7890

I'm trying to understand SIP, and one thing that keeps confusing me is phone numbers and dialing plans vs SIP URI's like me@domain.com. What is the difference? How do they coexist/work together? For ...
1
vote
1answer
337 views

Asterisk IVR to Extension slow

I have a IVR set to pickup our PSTN lines and tell the user to enter the extension OR to press 1 for a representative. If the person enters an extension it transfers within 1 second , but if they ...
1
vote
2answers
2k views

What is a SIP 'Gateway' and how is different from a SIP Proxy/Registrar?

Recently I started looking at SIP implementation for a future work. I was reading (Googling) about what SIP means and how to go about implementing a end-to-end SIP enabled VoIP network. What I did not ...
1
vote
1answer
450 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
1
vote
1answer
192 views

Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...
1
vote
1answer
198 views

What is SIP trunking?

Can someone explain to me in plain English what SIP trunking is, please? I've read about it on Google, but I don't really grasp it yet. Does it allow a VoIP call to be placed outside of the LAN? So ...
1
vote
4answers
3k views

Workaround for blocking SIP port (5060) by ISP

I found out my ISP is blocking outgoing SIP port (5060) at home. I have a remote Linux server that I can use to listen on different port than 5060 and do the forwarding for the traffic. Not sure what ...
1
vote
1answer
2k views

Asterisk and SIP behind NAT

I am trying to Setup an Asterisk-Server to accept calls from a client in an other Network. The Server and the client are behind an NAT. I have already activated STUN on the client, but I am still ...
1
vote
2answers
225 views

Can I set up an “involuntary” conference call with Freeswitch?

I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to ...
1
vote
1answer
67 views

Which one VoIP SIP audio codec should I choose to high quality calls?

The quality of VoIP calls depends on the codec used for the transmission and on the bandwidth of Internet connection. I want to use best optimized codec for my Internet connection. I have to select ...