Recommended VOIP Monitoring Tools

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As system administrator, monitoring the continuity of services is vital. Today I would like to highlight some tools that could come in handy for VoIP monitoring.

Nagios

For those of you who didn't know it, Nagios can be configured to monitor pretty much anything you want, including Asterisk servers. Actually, with Nagios the (much) harder part is deciding what’s relevant to monitor, and what your alarm thresholds should be set at. Some VoIP community members have reported that they used Nagios to monitor ~4,000 hosts and about 8,000 to 10,000 services before they started running into scaling problems on a single box. For…

Asterisk Tips 3.1 years ago 2 Answers

Flowroute: Howto Set Outbound Callerid (ast 1.4)?

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The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): "P-Asserted-Identity", "Remote-Party-ID" or "From:". I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44.

Asterisk Tips 3.1 years ago 10 Answers

Asterisk Directmedia

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What is directmedia?

"directmedia" is the new configuration option name for "canreinvite"; they are the same feature.

To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting is YES, for example in the SIP protocol configuration file sip.conf.…

Asterisk Tips 3.3 years ago 0 Answers

E & M signalling and Dahdi

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If you were asking if E&M over E1 signalling works on top of R2, ISDN and where can you find a sample Dahdi configuration?  Then, no it doesn’t. E&M signalling is the same layer as R2 and ISDN. It is an alternative to them, not another layer.

Asterisk Tips 3.3 years ago 0 Answers

Invite + decreasing sequence number => 500 Error?

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Consider the following  SIP conversation: C3 PBX to Asterisk

  • Case 1. Sequence Numer always increasing:
=> Invite
< = 401 Unauthenticated
=> Invite+auth with sequence number > previous Invite.
< = 100 Trying etc. Works OK.
 
  • Case 2. Sequence Number decreasing.
=> Invite
< = 401 Unauthenticated
=> Invite+auth with sequence number < previous Invite.
< = 500 ERROR
  It’s not a bug – decrementing the CSeq header field value is directly in violation of RFC 3261. From section 22.2: When a UAC resubmits a request with its credentials after receiving a 401 (Unauthorized) or 407 (Proxy Authentication Required) response, it MUST increment the…

Asterisk Tips 3.3 years ago 0 Answers

Asterisk Ulimit

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After re/starting Asterisk with a user other than root you might see this error:

"/usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted"
, even if you try to set the current ulimit to a higher level. Edit your '/usr/sbin/safe_asterisk' and comment (put #) the line 86 and 102. Or you can also modify '/etc/sudoers' in order to to allow your user to call ulimit via sudo. The reason for this is that Asterisk will limit the number of open files it uses if you are not a super-user, to 32K (look at line 32). Anyways, it's very likely that Linux

Asterisk Tips 3.5 years ago 0 Answers

SIP and NAT best practices in Asterisk

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What should I do in order to to be as secure as possible and with “clean” logs?

Well, for an article about Asterisk security best practices, consider reading this article. About SIP and NAT best practices, in short, the simplest answer is to always use ‘nat=yes’ (or at least ‘nat=force_rport’ in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you…

Asterisk Tips 3.5 years ago 0 Answers

Correct Format of US numbers

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If you were wondering about the correct format of US numbers then this might come in handy. In short: The proper way to send CallerID number is 10 digit (from NANPA numbers), no preceding 1 or plus sign (+) as this is handled by the provider to the end user if at all. How many numbers to send? First of all, if you are sending a CLID inside the US (Home NPA Local Calls) you should not send more than 10 digits (do not include the '1') or your call might be considered "Home NPA Toll Calls" or "Foreign NPA Toll…

General 3.5 years ago 0 Answers

No valid Transports Available, Falling Back To 'UDP'

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If after installing Asterisk 10.1.1 you see the following errors on SIP reload:  

WARNING[3665]: chan_sip.c:29242 reload_config: No valid transports available, falling back to 'udp'.

== Using SIP CoS mark 4
WARNING[3665]: chan_sip.c:27839 build_peer: 'tcp' is not a valid transport type when tcpenabled=no. If no other is specified, the defaults from general will be used.
  This might be because a possible regression. Please open an issue in JIRA referencing this thread. [1] http://svnview.digium.com/svn/asterisk?view=revision&revision=347727

Asterisk Tips 3.5 years ago 3 Answers

Outbound Call Load Balancing

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Due to high volume of outbound calls you might be asked to alternate the outbound traffic by your termination provider. Basically, what what you need to do here is: Having 2 destinations (Dst_A, Dst_B), check/set a variable in the AstDB. Then if the variable is 1, set it to 2 and route via Dst_A; otherwise, set it to 1 and route via Dst_B. This code snippet might be what  you are looking for:

GotoIf(${SET(DB(sw/provider)=$[!0${DB(sw/provider)}])}?Dst_A:Dst_B)
  Maybe in a future you might want to think in another approach to this task, but in the meantime this might come in handy.

Asterisk Tips 3.5 years ago 0 Answers