x-pro sip configuration Resources

X-pro Sip Configuration News

EMPTY feed http://voip-info-now.com/blog/index.php/feed/

Asterisk cmd FollowMe
FollowMe Synopsis Find-Me/Follow-Me application
Introduced with Asterisk 1.4, see patch 5574

Description  FollowMe(followmeid|options):

This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the <followmeid> voice over ip kenya in followme.conf. If the specified <followmeid> profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority.
Forked (=simultaneous) dialing of multiple numbers in the same step is supported with this application if you'd like to dial multiple numbers in the same followme step.

 Options:
   s    - Playback the incoming status message prior to starting the follow-me step(s)
   a    - Record the caller's name so it can be announced to the callee on each step
   n    - Playback the unreachable status message if we've run out of steps to reach the
          or the callee has elected not to be reachable.

Note that every change in followme.conf must be activated with a "reload app_followme.so" on the Asterisk CLI.

Important note: The app_followme that's in 1.4 right now do NOT make use of any assets in AstDB as described in the following lines.

The settings in followme.conf allow for an entry that points to the astdb, something like this:

 number => family/key
 number => family/key

Details Scenario Call comes in, outside caller dials "100" Desk phone for user Joe rings. No answer Joe's house phone rings. Joe's wife picks up and hears a voice "Please press any key to accept a call for extension 100." Joe's wife hangs up. Joe's cell phone rings. Joe picks up and hears a voice "Please press any phone service vonage news corp to accept a call for extension 100." Joe presses 1 and says "Hello this is Joe".
Alternately, in the penultimate step
Cell voice mail picks up. Voice says "Please press any key to accept a call for extension 100". No keys pressed since it's a voice mail Call is routed to Asterisk voicemail.
The a follow-me number to call is specified in followme.conf. You can specify as many of these numbers as you like. They will be dialed in the order that you specify them in the config file OR as specified with the order field on the number prompt. Therefore forked (parallel) dialing of multiple numbers in the same step is supported.

Often people want to force their 'follow me' to simply be on or off. So, when it's on, it doesn't ring their desk phone at all, just goes straight to cell/etc. When it's off, it only goes to desk phone, and nowhere else. You can achieve thisvia extensions. ... voip DFW

voip-info.org
Welcome to the VOIP Wiki - a reference guide to all things VOIP This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.

Notice: Editing pages to eliminate SPAM and inappropriate content is appreciated, when there are disagreements about what is appropriate content please contact: support@voip-info.org.
If you are removing content from pages, please leave a polite comment explaining the reason for the changes. Uncivil comments, user names, or content will result in the removal of the users editing privileges.


Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.


NEWS 2010-03-12 - VoIPon Brings the Best of CeBIT to You VoIPon Interviews Sangoma, AVM, Pika, Patton, snom, Grandstream 2010-03-11 - Announcing the 24-hour VoIPathon VoIP chat live to celebrate the three years of VoIP Users Conference 2010-03-11 - DTH Software, Inc releases new Express edition of its billing system for start-ups. 1-24 concurrent calls. 2010-03-10 - Apstel Visual Dialplan for Asterisk - Run Through with Screenshots. 2010-03-10 - Phone System Comparison Chart for Spring 2010 Released by CompareBusinessProducts.com - over 90 phone systems compared 2010-03-10 - VoIPon Interviews Rhino Equipment Corp. ... vonage wav advertisement

Asterisk v1.2 upgrade to v1.4 gotchas
The upgrade is fairly smooth, and there's lots of new features. It's the deprecated stuff that bites you during the migration process.

Start the migration with reading UPGRADE.txt, and then look at the CHANGES file for details that were introduced with 1.4.0.

Make sure to read the new xxx.conf.sample files. That way you may detect new features/options that not seldomly also fix potential security issues.

asterisk.conf For sure you will want to have "internal_timing=yes"!

extensions.conf Hurray, you may now monitor the call park and Meet conference with hint, use "Meetme:1234" or "park:701@parkedcalls"!
Call pickup has changed, in particular you really must take a look at PICKUPMARK.

A line starting with ;-- (semicolon immediately followed by two dashes) is now treated as opening a multi-line comment, so be aware! You might disable the entirety of what is remaining in your dialplan from this point on.

Changes to watch out for:
Calling a voicemail box with flags for busy or unavailable (options b and u) must now be performed with a pipe as opposed to prepending that option to the mailbox number: "b1234" or "u4567" turns into "1234|b" and "4567|u" SIP_HEADER() with (Via) now needs to be written as (Via,1) in Asterisk 1.4 LookupCIDName is deprecated. Please use the much more beautiful and easy-to-read Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) instead
Many similar changes for variables are described in ugprade.txt:
change ${TIMESTAMP} variable to ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} function change ${CALLERIDNUM} variable to ${CALLERID(num)} function

sip.conf In the [general] section "port=" has been renamed to "bindport=" to prevent misunderstandings The default for QoS settings has changed from the old TOS to the new DiffServ method. This also cheap voip long distance to Philippines to iax.conf, by the way. with the new subscribemwi=yes we can finally instruct Asterisk to not send what some SIP devices consider as unsolicited NOTIFY messages (AVM Fritz!Box, Siemens Gigaset and others). This prevents SIP ERROR 481 or "Remote host cannot match NOTIFY" BLF and hints you will need to set "call-limit=" to make hints (SIP SUBSCRIPTIONS) work in Asterisk 1.4 also look at the general setting "limitonpeers=yes" and "notifyringing=yes" etc.


iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up.

in general section, add: iaxthreadcount = 200 in general section, add: iaxmaxthreadcount = 1000
Later in 1.4.26.2 also this changed due to a security issue:

add this to iax.conf: calltokenoptional = 0.0.0.0/0.0.0.0
add