SIP Over 3G Mobile Network Using NAT

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It might be the case that you are are trying to use SIP client over 3G and It registers and call can be initiated from the client but it can't receive call; cause asterisk sever marks it as unreachable immediately after registration. Even more, the above works well when you turn off 3g and switch over to wifi. I recommend you to set "qualify=no", also check if your asterisk sip server Is available with ports on the public ip that your phone is trying to register from 3G network. Other issue might be SIP retransmision (no reply to our critical packet). You…

Asterisk Tips 10 months ago 2 Answers

Asterisk: Receiving Incoming SMS On Analog or ISDN Landline

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The way to tell if an incoming call is an SMS call or a voice call when receiving incoming SMS on an Analog or ISDN line in Asterisk is by reading the callerid. Normally there is only one Short Message Service Center (SMS Center) which can send you SMS on a fixed line. By looking at its callerId you can know if it is an SMS call. In the mobile world, nevertheless, we can not consider an SMS as a call, and the cellphone won't send a SMS directly to the landline phone. What it will do is to hand…

Asterisk Tips 1.6 years ago 0 Answers

Asterisk: How To Create A coredump

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In order to create a coredump in Asterisk you must compile it with “DEBUG_THREADS” and “DONT_OPTIMIZE” options turned on, remember to start it with: # /bin/bash /usr/sbin/safe_asterisk Se the Asterisk start script coredump-dir (DUMPDROP) as you prefer (the default is /tmp). Now, send a kill -6 signal to the Asterisk process. That should produce a coredump. Don't forget to read doc/backtrace.txt Thanks for reading.  

Asterisk Tips 1.6 years ago 0 Answers

Asterisk: Monitoring Your E1 or T1 With Nagios

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E1 monitoring/T1 monitoring is a need that we have on a daily basis because the client's business depends on it. Here you have an interesting script in PERL that will allow you to monitor your E1/T1 with Nagios. Passing the name of the port as an argument will allow you to use it even if it is not yet connected to a telco. Using the Nagios Remote Plugin Executor on the Asterisk server you can execute it. Additionally, it will report bad ports to you in order to avoid the guessing process.

 E1/T1 monitoring script in Asterisk

#!/usr/bin/perl…

Asterisk Tips 1.7 years ago 0 Answers

Unable To Create Channel Of Type 'SIP' (cause 20 - Subscriber Absent)

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I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed the official user manual and the blog post here http://www.skelleton.net/2012/08/02/linksys-spa-3102/

When I call an extension say 225 from the analog phone, I can get the IVR I have setup in my dialplan. But when I Call the analog phone extension using a sip phone I get the following error message:

"Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)" which usually means that the SIP device hasn't registered yet. If you are having RTP problems, get sure that uuid-devel and libuuid-devel…

Asterisk Tips 2.3 years ago 0 Answers

T38 Fax Detection Using g729 and Asterisk VoIP Software

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Using T.38 termination with Eutelia provider could present a scenario where you can receive faxes using g729, but not being able to receive it, even if you are able to receive it while using  alaw. This makes you to start suspecting of your VoIP Software. It might be the case that the remote endpoint, Eutelia, will need to detect the Fax Tones and send the T.38 ReINVITE to you, they may not have T.38 enabled. If that's the case, your Asterisk installation will need to detect the Fax Tones so as to make the decision the incoming call is a fax and then switch…

Asterisk Tips 2.6 years ago 0 Answers

Asterisk and OpenVPN + SIP configuration

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These are the instructions to configure OpenVPN + SIP configuration, based on a brainstorming discussion of the Asterisk Users Mailing List. The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. 1. Install OpenVPN on Asterisk server. On appliance, there’s only a single binary /bin/openvpn, and configuration files are in /etc/openvpn/. To be positive SIP/RTP packets go through the OpenVPN tunnel, make sure the firewall in front of the OpenVPN/Asterisk server only has OpenVPN port open (default: UDP 1194). 2. On client, from http://www.openvpn.net,…

Asterisk Tips 2.6 years ago 0 Answers

Asterisk Virtualization

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Last Updated: 29-Nov-2013 If you are about to dive into the process of Asterisk virtualization or are considering the options for any VoIP Software or PBX phone system then some primary issues might be preventing you from moving (e.g. the lack of proper timing as you need it for IAX2 trunking). If this is your case, consider these options and the pro and cons of each one:

  • OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant access to host node timing source (physical device, or dahdi_dummy…

    Asterisk Tips 2.7 years ago 0 Answers

Tips For Installing And Configuring Digum Cards

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What things do you wish you had been warned while configuring Digium Cards at client's site? These tips and suggestions will help you avoid pitfalls that will arise (according to Murphy's law) late in the night and when you have the bigger pressure to have the whole thing working. This information is based on a brainstorm that was raised in the community user's list.

Information you should ask for:

  • What are the Ts configuration settings? You could try to get an order form from the client.
    • Line coding (ami, b8zs, hdb3)
    • Framing (d4/esf, cas, ccs)
    • Jack type (RJ48X, RJ45)
    • ISDN…

      Asterisk Tips 2.8 years ago 0 Answers

Websockets On Asterisk 11 And SipML5

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In order to set up a user in Asterisk so that It can register via sipml5, on Asterisk 11. In the sip.conf entry for the account you are trying to register as place the following:

transport=ws
Example:
[3002]
username002
secret=XXXXXXXXX
host=dynamic
type=friend
context=test
disallow=all
allow=g729
;allow=all ; Allow codecs in order of preference
allow=ilbc
allow=silk8
allow=gsm
transport=ws
  Now, modify the sipml5 library so that the URL looks like this: ws://example.org:8088/ws (with the /ws at the end, as instructed). In order to get sure that the 8088…

Asterisk Tips 3 years ago 9 Answers