call comes in pstn gateway
Rings ring group
Extensions in the ring group have call forwarded to pstn numbers
The call should ring the pstn numbers
e.i: ring group test_rg has two extensions 1000 and 1001 and 1000 are registered from IP phone and in IP Phone I set call forwarding to my mobile number. When I call ring group it called 1000 and 1001. 1000 return 302 Temporary moved with new contact header with my mobile number. But FreeSWITCH is not processing 302 message and IP phone continues to send 302 until timeout.
I am facing this issue only when ring group strategy is the enterprise in all other it working fine.
Thank you in advance for your help.
There maybe a fix for this in the next major release 4.4 and already in the master branch. Reason this would may work is the Call Forward was completely rebuilt to prevent a user causing and infinite loop with their call forward.
When 4.4 is released follow the version upgrade instructions carefully. Make sure your delete your 'user exists' dialplan then go to Advanced -> Upgrade -> App Defaults for it to generate a new on for you.
Update 4.4 has been released as of 5 April 2018.
Related
Recently, we have had an increase in traffic to one of our servers with the user agent:
Mozilla/4.0+(compatible;+MSIE+8.0;+Windows+NT+6.1;+Trident/4.0)
This server receives click responses from emails, records them and then redirects to another server.
In theory, this user agent is IE 8 running on Windows 7 (32-bit), but the sudden uptick seems odd, especially given that Windows 7 shipped with IE 9.
As these are emails, I have looked through the Outlook user agents, but haven't found a match. Nor have I found any other mention of this user agent so far.
This traffic is coming from numerous IPs (mostly universities) and appears to be legitimate traffic from customers.
Is anyone aware of an email client, email server, firewall, etc. that may be sending this user agent?
If you never heard of IE 8 compatibility mode, now you should learn it,
https://learn.microsoft.com/en-us/previous-versions/windows/internet-explorer/ie-developer/compatibility/ms537503(v=vs.85)
That's exactly why you see MSIE 8.0. Many software (if they use IE control) might work under that mode, so you have very little control over that.
I need to check a list of about 40 000 mails if they are valid. I want to use this guide: http://www.labnol.org/software/verify-email-address/18220/
It works perfectly one by one but I am afraid that if I made a program that would check them all I could get banned mainly because it would be basically a dos attack.
What do you think? |Is there another way? I cant use any online service for that as I dont own the list of emails.
Thanks
I cant tell you if you will get banned but the techniques I would use to avoid getting banned are
Use a public IP address, proxy or vpn (eg mobile internet, wifi hotspot, TOR) as if you do get banned it wont effect you
start slow, process 5-10/s at first and then speed up, if you get blocked by one server go back to your last known good speed and don't connect to the blocked server for a while, check if you are still blocked manually
depending on the connection setup speed, only do a few emails per domain at a time. ie do 1 #gmail.com, 1 #Hotmail.com and 1 #yahoo.com per batch this stops you flooding one email server with thousands of requests at a time.
Hope this helps
I'm using Asterisk 11, a Cisco SPA303 Phone, and Twilio.
I can make outgoing phone calls without any issue and the call quality is top notch. On an incoming call however, my extension (and phone) ring, however when I answer the phone, there is no audio on either end and 30 seconds later, both calls end. Using Twilio's PCAP log, it shows that my asterisk server sends a BYE when answered. Asterisk however does not log a single thing on incoming calls. (All SIP traffic is logged on outgoing calls). Does anyone have any idea what's going on?
Update:
Turns out that I had my incoming extensions in the wrong context, once that was updated, incoming and outgoing calls worked without a hitch. I marked #arheops as the answer because the logging of unanswered calls lead to the diagnosing of the problem.
Very likly(but you not informed) your asterisk installation is after NAT.
IF so, you have configure asterisk as described in http://www.voip-info.org/wiki/index.php?page_id=410
Also may need change ports on router/disable SIP ALG on router etc.
Logging(you mean cdr,right?) is controled by /etc/asterisk/cdr.conf
Most likly you have ananswered=no in that file.
Hi Asterisk and VOIP Expert,
I want to setup a new VOIP business using asterisk. The requirement is as below:
Asterisk Server 1----------|
| My central
Asterisk Server 2----------| ----SIP Call Forward---> [Asterisk Server]--Trunk-->Call route to any Number
| Billing and Charging
Asterisk Server 3----------| +
CRM
can any one guide how to use my central asterisk server from other asterisk server and route to other part of the word. I mean call establish to any phone. For the call establish to out world..what SIP need to use? Or i need to install digisum cards on my central asterisk server or what?
Target is 20,000 Concurrent call. How concurrent call Asterisk 11.4 can handle?
Please cooperate for any information, or any link or diagram.
Usually you want to forward your outside calls to a VoIP provider (using SIP), so them can be routed to any number worldwide.
Take a look at http://www.voip-info.org/wiki/view/VOIP+Service+Providers
Digium cards, as interface cards of any other manufacturer (Sangoma, for example), could be used to connect your asterisk box to a digital or analog interface of a telecom. Which allows you to receive and send calls over.
First way is less expensive, but much unreliable. Second gives you solid business solution, but charges rise significantly when you start making long-distance calls.
You need to setup extension for each of your servers on central server.
After that you have use extension username/secret for trunk settings on other servers.
http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers
Unfortunately asterisk not able handle 20000 calls. Hi load servers(more then 200 calls) have be setup by experts, if more then 400 call - using other technologies like opensips(ser,kamailio)+rtprpoxy.
About asterisk hiload: fine tuned asterisk can handle 400-700 calls with media
Without media(just setups,media go directly) asterisk can handle 20-30cps on hi-end servers like 3.2Ghz 8-core xeons:
-- upto 4500 calls with ASR 50% and ACD 5minutes or less if ASR/ACD go down)
-- upto 1250 calls with typical ASR 25% and ACD 2 minutes.
-- only 20-30 calls when no success,so such setup can't be considered production worth.
Note, setup without media WILL have issues if both legs not trusted.
I have a Windows XMPP client - PSI and an android one - IMO. I'm connected to the same custom server, using two different resources (hostname on desktop, don't know what IMO uses as resource). When someone sends me a message, only desktop client is able to receive it. Android client can only send.
What to configure in clients to be able to receive messages on both clients simultaneously?
Figured it out. XMPP protocol has priorities assigned to resources. See 11.1 in http://xmpp.org/rfcs/rfc3921.html#rules. Valid range is -127 .. +128
IMO sends priority 1 (at least in my version). Setting priority in PSI to -120 made my phone client always receive the message. I'll play with priorities to take advantage of auto-away feature that lowers priority.
If you've got admin permissions on an Openfire server, setting the system property "route.all-resources" to "true" should allow all connected client to receive a message sent to a Jabber ID. This worked in my case.