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) ...
0
votes
1answer
17 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 ...
-3
votes
1answer
40 views
How do I configure my PBX? [closed]
I have two PBX, one has setup a sip trunk, but I have to get another trunk calls from the other pbx,
I have two Elastix PBX;
How do I configure my PBX "A" then my pbx calls "B" for the SIP trunk
...
-2
votes
0answers
44 views
Voip, what is the best opensourse solution? [closed]
I am in to networks for quit long time but never touched the subject of VOIP.
So what do you think would be the best solution with easy accessible web interface, for couple phones that can connect to ...
0
votes
1answer
52 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 ...
0
votes
0answers
55 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
91 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
57 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
53 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
75 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
112 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
157 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
83 views
SIP/Asterisk: Withheld number from client not set to withhold the number
I'm using PJSIP (a command line SIP client) to make calls on an Asterisk server (IP: 11.22.333.44).
When calling from the PJSIP client (number 12323) to another extension (12315, a Snom phone), that ...
0
votes
0answers
143 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
654 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
330 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
57 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
27 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
95 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 ...
2
votes
1answer
1k 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
197 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
98 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
0answers
343 views
Can't register SIP clients to asterisk11 server on Fedora [closed]
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
172 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
99 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
107 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 ...
1
vote
1answer
189 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
131 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
80 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
64 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
491 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
105 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
299 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
5k 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
2k 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 ...
2
votes
0answers
164 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 ...
0
votes
2answers
950 views
Asterisk: Authenticated SIP trunk
Let's say I have an Asterisk system with a bunch of connections: there are phones (who register itself with *) and providers (who wish to establish SIP trunks to put a lot of calls over, with ...
0
votes
1answer
549 views
VoIP Tunnel Implementation for SIP Client
I am planning to provide an option for tunneling in my SIP client. I have tried to search on web for open-source implementation of this, but couldn't find one.
My questions are:
1) If I go writing ...
1
vote
1answer
146 views
NAT setup for an SIP Registrar/Proxy
I am currently trying to get the following scenario to work.
Warning: I am a software engineer--not a network admin.
I have various SIP endpoints (sip based video servers) on a LAN. On a Windows ...
0
votes
1answer
94 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
4answers
293 views
Linux SIP Client for Server/Bot [closed]
I need a solution for just receive a call from a SIP Provider and answer the call automatically with a mp3/wav file.
I've already achieved this using a complex and big PBX solution like Freeswitch, ...
1
vote
3answers
944 views
Cannot connect softphone as a FreeSwitch Extension
Having successfully configured and maintained few Asterisk based installations, I have now been provided a task to configure FreeSwitch SIP server.
ISO downloaded from
...
1
vote
6answers
2k 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 ...
0
votes
1answer
409 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" ...
3
votes
2answers
984 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 ...
0
votes
2answers
2k 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 ...
1
vote
0answers
246 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 ...
6
votes
4answers
994 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 ...
1
vote
1answer
188 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 ...
2
votes
1answer
206 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
0
votes
1answer
235 views
how UAC should handle SIP 183 Session Progress
I have the following scenario:
2 UAC are trying to talk, via a remote SIP server (openSER/Kamailio 3.1.3) = client infrastructure.
The UAC software was developed over a local test infrastructure ...
