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 9 months ago 2 Answer

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 Answer

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 Answer

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 1.6 years ago 0 Answer

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 are installed. Also note that type=friend…

Asterisk Tips 2.2 years ago 0 Answer