3
votes
Accepted
SIP call connection is p2p?
The actual SIP protocol just negotiates the connection, and the actual call is transferred over RDP, usually directly between endpoints.
For a direct connection to work, they must be able to reach ...
- 3,357
3
votes
Register asterisk to sip trunk
You are missing two items to be placed directly under your register => entry.
[general]
register => myusername:[email protected]
registertimeout=20
registerattempts=0
As soon as you ...
- 31
3
votes
Register asterisk to sip trunk
First, the register line should have a path set at the end, like:
register => myusername:mypassword:[email protected]/84106639
Then do a sip reload or service asterisk restart. After ...
- 151
2
votes
Accepted
Why does scp congest my network?
This has nothing to do with SCP. If you performed a large file upload via HTTP, you would likely see the same issues.
Rather, it has everything to do with the fact that you're causing network ...
- 110k
2
votes
Asterisk SIP/2.0 401 Unauthorized
As this might helps for somebody:
insecure=invite
helped me after a similar relocation. For me it was vm to vm but bridged to the exact same network so I just don't get it why my working ...
- 41
2
votes
What is a SIP B2BUA, and how is it different from a UA?
You may find an interesting article explaining B2BUAs here:
https://andrewjprokop.wordpress.com/2013/12/27/the-back-to-back-user-agent-b2bua/
In a few words, a B2BUA is a SIP entity which can "split"...
- 21
2
votes
FAX on VoIP line does not work
Short answer: Use G.711 Codec and turn off T.38 support in your Router.
If that still fails, try lower baud rates in your Fax machine.
In the case of German Telekom, T.38 isn't supported. Your router ...
- 312
2
votes
Polycom SoundPoint IP 4000 won't power on. How to isolate power supply?
A Polycom SoundPoint IP 4000 SIP can be powered using ether power-adapter or a standard POE connection.
The VVX 601 phone should use the same power supply as the 4000 phone, as most more recent phones ...
- 209
2
votes
Hyper-V Virtual Switch not recieving any packets
PowerShell:
PS C:> Move-VM "VMNAME" "DestinationServerName" -IncludeStorage -DestinationStoragePath D:\"VMNAME"
Alternatively, you can do it with Windows 10 Pro machine and Hyper-V manager on it, ...
- 5,115
2
votes
Accepted
How to enable asterisk Call Completion on Busy Subscriber (CCBS)?
To have Asterisk enabling CCBS after a CallCompletionRequest the busy device need to answer the SIP INVITE with a 486 Busy Here response. That is, Asterisk have to dial the extension even if it have ...
- 36
2
votes
Accepted
Mikrotik SIP trunk routing
I just had to specify my DSTNAT rules for each ethernet port, instead of having using generic incoming rule
- 31
2
votes
Accepted
Configure siproxd to use behind a openvpn connection
As per siproxd FAQ ( http://siproxd.sourceforge.net/index.php?op=faq ) ensure that PC-Linux host has packet forwarding enabled ant that iptables is doing masquerading between the two nets.
If PC-...
2
votes
Forwarding SIP headers with asterisk (PJSIP)
You can use _VARIABLE: https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance
[handler]
exten => addheader,1,NoOp(Value is ${somevar})
same => n,Set(PJSIP_HEADER(add,X-myheader=${...
- 21
1
vote
Block FaceTime/Whatsapp video
Since Facetime and What app is encrypted traffic, It would not be possible to block
video unless the apps use a different URL for text messages and video. Knowing the port numbers does not help ...
- 1,180
1
vote
Capture the VOIP calls receive to a sip phone that send Notificantions to an API on a call
You can try tSIP softphone. There are few event types like "on call state" to which Lua scripts can be assigned. From Lua script you can use ShellExecute + curl or other similar utility.
If you need ...
- 131
1
vote
Accepted
Asterisk skips first DTMF
don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem.
I commented the line "disallow=all" on the trunk configuration. I kept using ...
1
vote
linux port forwarding between two nics
It seems like you are looking for this,
echo 1 > /proc/sys/net/ipv4/ip_forward
- 314
1
vote
Accepted
Using Asterisk as a gateway to Provider
Asterisk ⟷ FreeSWITCH
Your needs of course might be different but this is a good start—I have a couple servers with a private connection and so you may need to adapt authentication measures ...
- 144
1
vote
Asterisk client behind NAT - UDP registration does not work
When using Asterisk behind NAT, you should always add the externip option along with nat option to the [general] section of your sip.conf
Try this
[general]
nat=force_rport,comedia
externip=<your ...
- 3,725
1
vote
Call flow explanation
The call flow diagram is incorrect. Responses to BYE should be 2xx; ACK are only used to acknowledge responses to INVITE as mentioned. See RFC 3261 for more details.
At the end of the call, Bob ...
- 33
1
vote
Accepted
Getting PJSIP with TLS to work with Twilio SIP Trunking on FreePBX
After talking with Twilio support, encrypted SIP trunking is only supported on PJSIP 2.5.5 or higher. I tested it on an Alpha build of the FreePBX Distro which runs 2.5.5, and it still complained ...
- 497
1
vote
Accepted
Connect Asterisk system to SIP provider with Asterisk dynamic realtime
The comments in the default version of /etc/asterisk/sip.conf pointed to a solution. An entry of the form
register => user:[email protected]
is apparently equivalent to a [device] ...
- 706
1
vote
freeswitch received audio delayed for 20-30 seconds
what's the network topology between the phones and FreeSWITCH server? Maybe there's a firewall that is doing NAT or SIP intelligence. Also if you run packet captures, it would help to determine where ...
1
vote
FAX on VoIP line does not work
If the carrier won't support t.38 you'll have issues, most don't and act like you are crazy for asking. g.711 can work but will probably give you headaches.
Best solution is get an analog line for ...
- 486
1
vote
Hacker bypassing iptables
The only plausible explanation how it could work is, that the offending UDP datagrams somehow pass --state ESTABLISHED,RELATED.
Given that UDP is a stateless protocol, I doubt that the state module ...
- 2,060
1
vote
How can I simulate a modem over a VOIP connection?
When this question was asked, there was no existing solution, but Vocal Technologies has a SAMS (SIP Analog Modem Server) which is what the OP was looking for. It is a purely software modem server ...
- 11
1
vote
SIP.js connection to Asterisk 11.20 over WSS not working
I was using self signed certs. self signed certs are not considered trustworthy. Switching over to valid SSL certs did the trick. Chrome 47 and higher will require this.
- 11
1
vote
Accepted
SIP: No matching Codecs / 480 Temporarily not available
The sad truth is, those non-elmeg system phones didn't register correctly.
I had to assign PINs for the Users in the Hybird 130j Interface, which is then used to authenticate with the System.
Elmeg ...
- 312
Only top scored, non community-wiki answers of a minimum length are eligible