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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636 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
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1answer
495 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
285 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 - ...
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1answer
189 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 ...
1
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1answer
363 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 ...
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1answer
182 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 ...
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1answer
28 views

ASA is not forwarding packets to egress interface

We are seeing some very strange behaviour from our Cisco ASA 5505 running 9.1(2) We have a SIP PBX inside our network. It's got a bit of an odd configuration, it listens for inbound SIP requests for ...
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1answer
113 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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1answer
197 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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votes
1answer
599 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 ...
0
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1answer
426 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 ...
0
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1answer
46 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
votes
1answer
112 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 ...
0
votes
1answer
118 views

SRV records not populating correctly

I have a lync box with a SIP SRV record that isn't working for a significant portion of clients, despite their ability to resolve the FQDN of the edge server. What could cause inconsistent resolution ...
0
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1answer
456 views

Avaya 9630 for SIP

Is there any way to make the 9630 Desktop Headset work just as a SIP IP telephone, without the extensive need for additional Avaya equipment. All of our infrastructure is Cisco and we may receive ...
0
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1answer
349 views

problem with red5phone unable to connect

i'm looking at red5 and red5phone to make it works. i am getting an error netconnection: Status: 1 rtmpconnection #phone _onStatus NetConnection.Connect.Failed netconnection: Status: 0 connection ...
2
votes
0answers
106 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
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0answers
55 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
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0answers
282 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
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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
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0answers
425 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, ...
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0answers
134 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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0answers
185 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 ...
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0answers
112 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?
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465 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 ...
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0answers
321 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 ...
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0answers
41 views

add presence for a remote user to a legacy telephone system?

we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at ...
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0answers
11 views

Asterisks Dialplan Extensions

I'm playing around with a very simple asterisks setup. My asterisks server is 192.168.1.10 (Ubuntu 12.04), my 2 sip clients are 192.168.1.20 and .21 (both using ubuntu 12.04 and Jitsi as a sip ...
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0answers
11 views

SIP UAC crash during a call

If a SIP UAC crashes in the middle of a SIP call, what is the expected behavior of the UAS and the remote UAC? Assume that a single UAS connects both UACs in the call. Is this scenario discussed in a ...
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0answers
133 views

How to set up kamailio proxy server and route calls to Twilio?

i am trying to route all calls to twilio through kamailio proxy. with my config file, call gets connected and automatically drops after about 30 seconds. This is because ACK sent to twilio for 200 OK ...
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0answers
33 views

Kamailio doesn't restart with snmpstats module

I have Kamailio 4.2.0 running in a RedHat Entreprise Linux Server and I wanted to add some KAMAILIO-MIBS in order to monitor the machine. I went to the ...
0
votes
0answers
154 views

Asterisk 11.* TLS Configuration

I have installed asterisk 11.12.0 on CentOS (64-bit). When enabling the TLS support i get the following error WARNING[7620]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! i tried ...
0
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0answers
115 views

VLAN setup Cisco SG300 and SIP gateway

Hopefully someone can shed some light here. I'm quite new to networking and not too sure if I'm doing the right thing here. We are setting up a new VOIP phone system with SIP gateway for our company. ...
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0answers
41 views

Asterisk SIP calls between remote L2L VPN sites

I am running into an issue trying to set up another remote L2L VPN site with VoIP running SIP and Asterisk. Each remote Site (site 1 and site 2) Can talk to the Asterisk server and any phones on the ...
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0answers
144 views

Dual WAN setup in Redhat/Centos Linux

I want to setup a Linux Server running SIP (freeswitch) applications with 3 NICs as follows: 2 of the NICs are WAN connections and the 3rd is a Private network for routing requests to other hosts on ...
0
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0answers
94 views

sip webrtc call to softphone 488 Not Acceptable Here

I'am using resiprocate stack on the server, it's working fine for softphone-softphone, browser-browser and softphone-browser calls, but when I'm trying to call from browser to softphone it failes with ...
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0answers
48 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
63 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 ...
0
votes
0answers
60 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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0answers
25 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 ...
0
votes
0answers
150 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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0answers
133 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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votes
0answers
64 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
72 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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0answers
94 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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0answers
367 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 - ...
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0answers
614 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 ...
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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 ...
0
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0answers
779 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 ...
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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 ...