similar to: Changing volume via dialplan

Hello, I've recently setup a small load test against an instance of Asterisks. I've tested on asterisk 13.5 and 14.6 with the same results. I am using PJSIP. My dial plan is, [test] exten => 1001,1,Answer exten => 1001,n,MusicOnHold(15) exten => 1001,n,Hangup I am using SIPP to test. I can share XML if desired but it simply waits on the line while music plays for 8

Hi Richard, Thank you for the reply Correct, I did mean 13.15. I set no optimize and better backtrace through "make menuselect" and the output is now [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0) Got 26 backtrace records #0: [0x61923f] main/utils.c:2475

Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -------------- next part -------------- An HTML attachment was

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,

Greetings all, After a long day of fighting with GTalk and having it finally working, I wanted to setup DPMA on my Digium phone. So first of all, I had to reinstall it all and reconfigure it all, since it works only on certified versions, and my installation was not from the certified branch. It took a long time of recompiling, testing, adding missing stuff, but I got it straight. Now, I

Hello, I've recently given a try to a Digium D70 phone. At the moment, I'm configuring them though config files with a DHCP server and not using DPMA. Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0). Which is the best place to: - read about past issues - open new tickets for remaining issues. Best regards -------------- next part -------------- An HTML attachment

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

Is there a reasonably easy way to increase the volume on a DAHDI channel? The VOIP phones in the house work OK, but for the phones connected to DAHDI channels on a Digium TDM400P card, the volume is very low and it's hard to hear if there is any background noise at all. If this is documented, point me to where and I'll gladly do my reading. Thanks, --Greg

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have just managed to get my 1st * server up and running and have a lot of issues with theX100P analog card. Would really appreciate anyone trying to help me on the following : 1. The receive and transmit is too soft. So i increased the txgain and rxgain. The volume is fine after this, but there is a lot of 'wind' noise on the line. I have my echo cancellation on, aggregive

Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? maxx

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten

I still have problems with beeps and clicks on all my calls. I have polycom sip phones. I also can hear the beeps and clicks on some of my messages, which would lead me to believe that it is more of a decoding problem on the zaptel card. Any ideas? Thanks Sean Garland

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

For those that have experienced low VM recording volumes when using a Digium TDM04b (or similar analog pstn card), a work around has been committed to cvs-head. Need some folks to test it; it doesn't seem to work for me, but need some feedback from others to ensure the work around is actually functioning. (The work around relates back to bug #2023, and was committed around Thursday or Friday

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten