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) ...

learn more… | top users | synonyms

0
votes
1answer
211 views

Kamailio to accept anonymous registrations/custom registration authorization? [closed]

As a minimum requirement, I need Kamailio to accept any login with an empty password. Ideally, I need to perform some simple validation of the login in a script (Lua/Python), and save the given login ...
0
votes
2answers
58 views

Explanation SIP technology

I have a question regarding SIP. Well, whether on a regular VPS I am capable to put a workable server? I wanted to play with voice communication on the Android platform, I read that it supports the ...
0
votes
2answers
41 views

SIP Unregister: tag value

Does the SIP un-register tag value (in the From: field) should match the register tag value? i.e. if: REGISTER sip:registrar.biloxi.com SIP/2.0 Via: SIP/2.0/UDP ...
0
votes
6answers
8k views

How can I stop SipVicious ('friendly-scanner') from flooding my SIP server?

I run an SIP server which listens on UDP port 5060, and needs to accept authenticated requests from the public Internet. The problem is that occasionally it gets picked up by people scanning for SIP ...
0
votes
1answer
180 views

Asterisk Register username with special character like “@”

I am using a SIP provider that has provided me with a username like: +1122233344@aaa.bb.com (Note this is only the username part) And has a numerical password. My Register string looks something ...
1
vote
2answers
269 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 ...
2
votes
0answers
59 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?
1
vote
0answers
119 views

How can I use the FLASH key of my IP phone to transfer a call (atxfer)

My IP phone has a FLASH key. This key sends the DTMF of Flash [16 in decimal] (http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling). How do I set the atxfer parameter in the features.conf?
1
vote
1answer
69 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 ...
1
vote
1answer
242 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
0
votes
1answer
48 views

Advice respect an architecture PBX [closed]

i'm new on the world of PBX and network infrastructure. I worked in a model and wanted to know how feasible it is to implement it in a production environment. The requirement is to create a solution ...
0
votes
0answers
674 views

Avaya-Asterisk call drops immediatly via SIP (sip:anonymous@anonymous.invalid)

I'm trying to call from Avaya to Asterisk (11.5.0) in SIP. Avaya calls to Asterisk, then Asterisk rings my softphone but when I answer the call (in the softphone), call drops immediately. This is the ...
0
votes
0answers
35 views

Routing SIP client thru WAN router

I may ask again: I have following setup: SIP Client on LAN can make calls, etc thru PBX (is a closed device). I want the same for the remote SIP client. Can I configure all in the WAN router ? May ...
1
vote
1answer
268 views

Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...
0
votes
2answers
3k views

How to install Asterisk with SRTP support?

I'm rather new to Asterisk, and I need my server to support WebRTC. As far as I know, Asterisk version in Asterisk Now is compiled without SRTP support, which is necessary for WebRTC. So, I try to ...
1
vote
3answers
438 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
0
votes
1answer
85 views

AT&T Business Class Internet Router Options [closed]

I have a situation where AT&T seems to be very uncooperative in their selection of modems (routers) for their business class DSL service. They essentially gave us only two options for modems ...
1
vote
1answer
1k views

VoIP Network Components: Difference between Signaling Gateway, Media Gateway, Softswitch [closed]

In my degree course of Software Engineering we had a lecture in Computer Networking about VoIP and the Session Initiation Protocol (SIP). There were many terms brought up for some network components ...
1
vote
2answers
176 views

What determines the length of RTP packets?

I have established a SIP session between two clients. I observed the RTP trace between them on wireshark. The 'length' column has a value of 172 for the RTP packets that flow from Client 1 to Client 2 ...
0
votes
1answer
577 views

Asterisk trunk SIP

I have a problem with trunk SIP when I connect my Asterisk to my provider and the Internet connection is lost, the trunk SIP will be unreachable, the problem consist is all local extension are ...
0
votes
1answer
185 views

SIP (G.711) packets getting out of order

I have 2 asterisk servers on 2 different datacenters. Call is going from one to another. Suddenly audio quality decreased. I have made SIP traces on both servers. Here's what I've found: Server A ...
1
vote
1answer
331 views

What is the best option for SIP service redundancy?

From your practical experience, what is the best option for SIP service redundancy? a) one SRV record pointing to a hostname which has two A records b) two SRV records pointing to two different ...
1
vote
1answer
385 views

Serious Packet Loss on UDP using Amazon EC2

I am using ec2 to host a udp TURN server for SIP purpose. Anyway I get a lot of packet loss during voice call, so I did a udp performance test using iperf util. Result: [ ID] Interval Transfer ...
1
vote
2answers
398 views

VoIP client permamently losing connection after ISP IP reset

I have a SIP client device connected to my ADSL router, so it is behind a symmetric NAT. My ISP resets the client's IP every 24 hours, it looks like this in the router log: Mar 26 12:36:56 daemon ...
0
votes
1answer
222 views

Custom ENUM/NAPTR DNS Server

I need to configure custom DNS server to handle ENUM requests for VoIP calls I have custom built DNS server in Java that handles ENUM requests for my company's VoIP services. ENUM is used 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. ...
0
votes
1answer
232 views

Troubleshooting a failing SIP connection

I have a linux Asterisk server that is behind a firewall. The firewall has three interfaces : eth0 is the LAN, where Asterisk resides, eth2 is the default gateway to the internet (via an ADSL modem ...
2
votes
1answer
670 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 ...
0
votes
1answer
444 views

Networking issue with asterisk server

I replaced an IVR machine for incoming call after going down. It is running asterisk 1.4.23 on ubunutu 10.04 I decided to put the server behind iptables because my server was under brute force attack. ...
0
votes
0answers
815 views

Not able to register via SIPclient on asterisk server?

I'm unable to register a SIP client on my Asterisk server. What changes do I need to make to my Asterisk server configuration to fix this? Here are my configuration files for Asterisk. sip.conf ...
0
votes
1answer
2k views

How to route/forward/connect incoming SIP call through other SIP provider to a phone number?

I have 2 VOIP SIP accounts. One is for a landline (incoming). The other should be purely outgoing and allows for calls to my mobile phone. Basically I want to connect the landline to my mobile phone. ...
0
votes
1answer
1k views

Fortigate restrict traffic through one external IP

I've got a fortigate 400A at a client's site. They've got a /26 from British Telecom, and we're using 4 of those IPs as a NAT Pool. Is there a way to say that traffic from 172.18.4.40-45 can only ...
2
votes
2answers
118 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 ...
0
votes
0answers
45 views

Tool to remove irrelevant sip messages from debug

I have a trace file of a cisco ccsip messages debug. Since this is taken on production, a lot of call that don't interest me are included. I want te easily filter out only one specific call-id, and ...
1
vote
1answer
483 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 ...
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 ...
0
votes
1answer
926 views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
0
votes
1answer
229 views

can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) ...
0
votes
2answers
781 views

Why are SIP calls via my server silent?

I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually). I can connect to it, make a call to other endpoint (namely, my android ...
1
vote
1answer
216 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 ...
0
votes
1answer
454 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
2
votes
1answer
516 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 ...
1
vote
1answer
229 views

LYNC - SIP “413 - Request entity too large” error to SUBSCRIBE requests

I'm currently trying to federate two Lync domains with one custom SIP proxy in the middle. The problem is that the front end server will reply "413 - Entity too large" to subscribe requests coming ...
1
vote
1answer
115 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 ...
0
votes
2answers
71 views

What do I need to go from having a domain to being able to take a sipuser@mydomain.com call?

(Sorry if this question is a little basic or should go on a different SE site. Just let me know.) I'm trying to learn about SIP. The problem is that I'm at the point where I don't know enough to try ...
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 ...
1
vote
1answer
188 views

Routing SIP between subnets

I've got the following configuration : 2 subnets (we'll call A and B) with IP phones in both. Internet access behind a NAT for each of them, and a routed VPN is set up between those 2 subnets. I am ...
1
vote
0answers
475 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic ...
3
votes
4answers
11k 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' - ...
3
votes
1answer
6k 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 ...