Questions tagged [sip]
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) sessions. Sessions may consist of one or several media streams.
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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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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 /path/to/...
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2
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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 ...
sunil221
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9
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How can I check whether port 5060 is open in centos? [closed]
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 ...
tawfiq
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2
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Hacker bypassing iptables
(moved from SO)
I have iptables protecting a sip server. It blocks all IPs except ones I specifically opened, and it seems to work for almost everyone. I have tested from lots of ip addresses that ...
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7
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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 ...
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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 ...
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5
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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 ...
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WebRTC on standalone asterisk - no audio
After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows:
Im not getting audio from WebRTC to WebRTC clients. I work in a LAN ...
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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 ...
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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 ...
- 1,991
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3
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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 ...
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What is "Cisco STG" and why would it dynamically replace a wildcard certificate on port 5061?
I have a lync client that is connecting to a Lync Edge server on port 5061. I get an invalid certificate error when connecting.
When I run wireshark, during the TLS setup, and inside the certificate ...
- 8,963
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4
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Register asterisk to sip trunk
I want to register my asterisk server to a SIP trunk. I have added following piece of code in my sip.conf and extensions.conf
sip.conf
[general]
register => myusername:[email protected]....
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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 ...
Gary B2312321321
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4
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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 ...
- 249
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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 ...
- 30.2k
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1
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Asterisk behind NAT sets wrong Contact Header
I'm using SIP with asterisk 13.1.0 behind a statically configured NAT.
The servers private_ip differs from the public_ip, where I can reach it.
I've already set these options in the sip.conf file.
...
4
votes
4
answers
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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:[email protected]>' failed for '37.75.210.177' - ...
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0
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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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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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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
http://wiki.fusionpbx.com/...
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Asterisk Register username with special character like "@"
I am using a SIP provider that has provided me with a username like:
[email protected] (Note this is only the username part)
And has a numerical password. My Register string looks something ...
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2
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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 call-...
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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 ...
user74078
3
votes
1
answer
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Hyper-V Virtual Switch not recieving any packets
I have a virtual machine set up with two network cards, one card is connected to a virtual switch for connection to the main network, the second is connected to another external port on the host ...
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Asterisk: execute shell command on server when call is accepted on SIP extension
I'm trying to configure asterisk to execute shell command for incoming calls - but only when the call is accepted. I've managed to setup extensions.conf so that command is executed when new call comes ...
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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 ...
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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 ...
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1
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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 ...
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3
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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 ...
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1
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Asterisk skips first DTMF
I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I ...
3
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2
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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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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 ...
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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 ...
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2
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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 ...
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0
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Asterisk times out and terminates connection after 6400ms due to 'no response'
I have a SIP trunk set up with Twilio for outbound calls. Twilio-FreePBX and then my test device is the simple X-Lite from CounterPath.
I can make an outgoing call from X-Lite. My cell phone rings ...
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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 ...
- 12.1k
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votes
2
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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 ...
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nmap results to find open ports for SIP
I suspect that a firewall, or other security, on either the router, or on tleilax or doge is causing a problem with SIP calls. How do I establish that the connection is allowed and not being blocked?
...
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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 ...
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2
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1
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Virtual PBX for automated attendant and SIP clients [closed]
I have a task to make our own virtual PBX for automated attendant and SIP clients.
Scenario is following: We have our SIP account given from our voice carrier now I want to share these lines with ...
- 1,531
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3
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FAX on VoIP line does not work
My provider switched me to an IP-based line. Now my analog ISDN (G3) Fax does not work anymore.
Is there any way I can use the conventional Fax with a SIP connection?
My Router supports T.38 Fax ...
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2
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Asterisk WaitForSilence NEVER detects silence
I am trying to use my dialplan to play recordings with WaitForSilence to make sure it wait until the person is done speaking or the message is left on voicemail. However, it doesn't seem to wait for 5 ...
2
votes
2
answers
154
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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 ...
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1
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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
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What is a SIP B2BUA, and how is it different from a UA?
I've been reading some stuff SIP related, and I'm confused, about what is a SIP back-to-back user agent(B2BUA) and a 'normal' user agent?
Can anyone explain the differences?
From what I read I can'...
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4
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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 ...
Ahmad Al-Ibrahim
2
votes
1
answer
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How to enable asterisk Call Completion on Busy Subscriber (CCBS)?
I cannot enable Call Completion on Busy Subscriber (CCBS) on asterisk, witch is part of the Call Completion Supplementary Services (CCSS), as is the Call Completion on No Response (CCNR) feature.
...
2
votes
2
answers
15k
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The benefit of different SIP and SMTP domains?
I was wondering if there was an advantage of having a different Lync SIP domain name from the Exchange SMTP domain name?
I could see the disadvantages, but I'm trying to understand the real world ...
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