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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946 views

Sip configuration for Elastix, for Terrasip (Provider) with Multiple DIDs

I have just purchased several DIDs with Terrasip I have configured my Elastix (freepbx) SIP trunk according to their suggestion: Please follow this template configuration. (valid for outbound/inbound ...
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5answers
3k 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 ...
2
votes
0answers
28 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 ...
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1answer
1k views

Asterisk 401 Unauthorized when trying to register sip clients

Trying to register a sip client to my asterisk server often (just about 90% of the times, not always, weirdly) results in 401 Unauthorized errors. This is the config for one of the extensions: [11] ...
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0answers
23 views

OpenSIPs Ubuntu 14.04 step by step installation and configuration [closed]

Can someone who has installed OpenSIPs give me a step by step instruction on how to install it on Ubuntu 14.04. I need to install it and setup a database. In the end of the installation, I need to be ...
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4answers
4k 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 ...
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0answers
34 views

Snom sip phone strange behavior

Didn't know where to ask this question, if someone can point me out the correct place, i'll be grateful I will be grateful, if someone will be able to help me with the following, I have around 200 ...
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0answers
14 views

Combining Elastix And Zultys PABX

have anyone integrated Zultys MX250 with Elastix? I got some problem with it. Before, my PABX is Elastix, now i want to upgrade using MX250 but keep Elastix as gateway for MX250. But i can't call out ...
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2answers
59 views

Allow specific SIP connection FreePBX

Hey I am using FreePBX with firmware: 5.211.65-14 and service pack 1.0.0.0 On this asterisk server I have everything up and running, but inbound phone calls might be rejected: ...
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8answers
11k views

One way audio on SIP phone on one call direction.. where to look next?

Got 2 SIP gateways (ZultysMX250). Sip trunk between them, they are in separate locations, different subnets 192.168.XX and 192.168.YY connected by IPSec VPN. Some phones work perfectly. But mine... ...
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votes
2answers
870 views

Lync Server 2010 with Hosted VoIP PBX

We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is ...
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votes
1answer
56 views

route incoming asterisk sip calls - fake auth rejected

we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?) what should happen if it works: user calls ...
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1answer
106 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 ...
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1answer
54 views

sip unreachable after changing routes using linux router

SIP server was unreachable after changing routes. I have 2 ISP ,ISP 1 is my default and ISP 2 is my back-up. I have a dialer behind linux router linux. Linux router has 3 ethernet cards. Router ...
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0answers
26 views

SIP/2.0 400 Sip message body is shorter that is indicated by the content length

During SIP communicating between my application, Freeswitch, and nextiva trunk, I am getting the following error message from 208.73.146.95 (Nextiva server): SIP/2.0 400 Sip message body is shorter ...
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1answer
769 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" ...
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votes
2answers
48 views

Asterisk inbound trunk unauthorised without allowguest=yes

I am trying to configure an Asterisk (Elastix) box to receive SIP calls from a provider without requiring allowguest=yes to be enabled in sip.conf. Basically the SIP trunk provider uses multiple IPs ...
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2answers
64 views

SecAst: Failing to Ban IP ''

To Generation D: This was the setup at time this issue was observed: secast-1.0.1.0-x86_64-ub12 on Ubuntu 12.04.4 Server LTS with Asterisk 11.10.2. The following events were captured and observed in ...
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1answer
119 views

SIP INVITE packet has WAN address rather than call manager LAN IP

I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN. I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 ...
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0answers
16 views

udp not working with sipp

I am doing some end to end testing using SIPP on RHEL 4.2. Both my client and server machiens are running RHEL 4.2. When I try to issue the command : sipp -t tn -sn uac <server_vip> -i ...
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vote
1answer
134 views

Freeswitch : Audio handshake failure 1 error while making call using Sipml5

I just installed FreeSwitch and successfully connected to server with user 1001. Details -> OS - Ubuntu 12.04 LTS 64 bits FS - 1.5.13b+git~20140614T114905Z~fc7a74905b~64bit OpenSSL - 1.0.1 chrome - ...
13
votes
1answer
1k 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 ...
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0answers
75 views

Iperf UDP listening on used port

I get WARNING[9157] chan_sip.c: Retransmission timeout reached on transmission error on asterisk and trying to troubleshoot the problem. Now asterisk is listening on 5060 port: udp 0 0 ...
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votes
1answer
184 views

Asterisk server running on loopback address instead of local ipv4 address

I installed Asterisk on my local Ubuntu 12.04 machine. After starting the asterisk [asterisk -c] I ran the command netstat -nlpa | grep asterisk and the output shown is this -> tcp 0 ...
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2answers
1k 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 ...
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0answers
42 views

Homogeneous cluster versus heterogeneous cluster

By homogeneous cluster I mean a cluster of nodes that all have the same services running. I am trying to set up a cluster of nodes that offer a set of services. The cluster can respond to HTTPS ...
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0answers
36 views

How do I make Cisco phones use <user>@<domain> instead of <user>@<ip> in SIP messages

I am using Cisco 8000 and 9000 series IP phones and registering them to Opensips. The problem is that I want to communicate with the proxy using @ instead of @ so I can have multiple domains on one ...
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0answers
332 views

Asterisk Real time does not perform register with SIP TRUNK provider

I am using astersik real time (dynamic). I have entered my SIP trunk details into the sippeers tables. However, the sip trunk does not perform a register with the SIP TRUNK providers servers as it ...
1
vote
0answers
83 views

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my ...
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vote
2answers
669 views

SIP Server with dynamic public IP address and dyndns

Is it possible to run SIP server with dynamic public IP address and dyndns? The question is what happens when IP address is changed and DNS is not yet updated.
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votes
1answer
71 views

Normal phone number over ethernet, what is this service called? [closed]

Plug in phone to wall with ethernet cable Someone calls a toll free number Your phone rings What is the correct, vendor-neutral term for this service? Some terms out there are: Comcast calls ...
3
votes
2answers
5k 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
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0answers
57 views

Connect Cisco IP phone on two Asterisk sip servers

I have a cisco ip phone with sip firmware v9.3.1. I want to connect two lines on two seperate Asterisk sip servers. In section i can only define USECALLMANAGER (in node) and not the IP of an ...
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votes
0answers
264 views

Configure SIP Trunk from Lync 2013 to Cisco Call Manager v6

I'm implementing Enterprise Voice for Lync, and I have to get Lync to work with a antiquated Cisco Call Manager version 6. The SIP trunk has been created but I get this error on my Lync servers: A ...
1
vote
0answers
79 views

freeswitch forbidden gateway error

i've stuck into a strange problem, i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX ...
0
votes
1answer
55 views

Cannot access Asterisk sip from remote location

I've used this tutorial to configure my iptables so I can access asterisk sip from outside of my server. My iptables -L looks like this Chain INPUT (policy ACCEPT) target prot opt source ...
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0answers
61 views

opensips sever running fine and UDP connection cannot register but TCP connections works fine

Im a newbie to opensips and Centos/Linux. Recently installed opensips sever which running smoothly on TCP connections, but users cannot register on UDP connections. here my Log, have a look. Feb 26 ...
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votes
0answers
177 views

Soft phone not getting register to my asterisk server

I have config in my local lan asterisk for test. which is working fine. With the same OS version and packages on "AWS Cloud" I am setting as following. But it don't get register "error = sip 408 - ...
0
votes
1answer
801 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 ...
0
votes
1answer
43 views

Authentication for SIP

I am reading about authentication protocols for SIP. I found that all protocols transfer the id/Username during the mutual authentication phase, directly with the message. Is it necessary to have the ...
0
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0answers
280 views

Asterisk: Retransmitting again and again

I have following sip settings Name : external Description : Realtime peer: No Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : public ...
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vote
1answer
107 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
0
votes
1answer
2k views

Cisco ASA 5505 - NAT or Port Forward for SIP / VoIP ver 8.4

I just had an NEC PBX installed that lets me use SIP trunks for VoIP services, My gateway is a Cisco ASA 5505 running 8.4 and I only have one public/static IP Addresses. So far, my trunks are ...
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2answers
158 views

asterisk - use a different “SIP CoS mark”

On my asterisk server, when i do a sip reload, i get the message "Using SIP CoS mark 4", followed by a registration time out. I need to make that statement say "Using SIP CoS mark 5". How do I ...
0
votes
1answer
114 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 ...
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2answers
41 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 ...
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vote
1answer
299 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 ...
4
votes
5answers
2k 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 ...
1
vote
1answer
1k views

Asterisk, IAXModem & Hylafax how-tos?

I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having ...
0
votes
2answers
31 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 ...