similar to: 1.2.x -> 1.4.x upgrade: dialplan block no longer works

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hello, After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I no longer match any extension. Apparently the "dad" is empty. However I can see the number just before it (146472130): P[ 4] I IND :SETUP oad:!?145201798p ?146472130 dad: ?146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: Extension can

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

I've have a simple caller id lookup on incoming: [teliax-in] .......... exten =>s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) ................ [set-callerid-name] exten => 0,1,NoOp( no CALLERID num set) exten => 02135590993,1,Set(CALLERID(name)=Matthew ) ............................................... exten => _0!,n,NoOp(CALLERID: ${CALLERID(name)}) exten => _0!,n,Return()

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Having purchased a license for 5 g729 channels on Digium's web shop I thought registration and installation would be a snap. NOT. I followed registration instructions to the letter but it failed with that message: ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server Which is stupid as my * box is a firewall and sits

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

Hello, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! What is then the recommended substitution for Gosub() application

How are we all documenting complex dialplan? Is there something similar to Doxygen? I've got around 20 config files covering around 60 contexts and 40 variables. Of course, I've maintained a basic list of the major stuff, and documented the code throughout, but it's grown to the stage where it needs to be better documented, have a proper flowchart etc. Talking of flowcharts, I see

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, In our backup script we sometimes would like to override the common (i.e: static) excludes filter list. For example we exclude "- *.ext" for all backups but would like to include "+ *.ext" only for 'local' backups. Are such entries supposed to cancel each other? How can one override an earlier exclude in a filter list? Thanks,

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Is there a way to have markdown automatically convert obvious (http, mailto) URL's to links? i.e: http://example.com -> <a href="http://example.com>http://example.com</a> Thanks, -- http://www.critikart.net

Hi, If several phones register to the same sip.conf section what will happen with a "Dial SIP/shared" in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy!

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? Kind regards, Jonas. -------------- next part

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context: