similar to: Asterisk 1.6 Overlap dialling timeout?

Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only

I've been having trouble matching variable extensions on a zap channel (an E1 line). Doing it the extensions.conf way: [pri1] ; Match 8078078- calls include => m807nat include => m807mob include => m807oth [m807nat] exten => _80780782XXXXXXXXX,1,StripMSD(7) exten => _2XXXXXXXXX,1,SetVar,clidest=${EXTEN} exten => _2XXXXXXXXX,2,Goto(cli,s,1) [m807mob] exten =>

Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the "Saved useragent...for peer 111" line appears on the Asterisk console, then the 111 user can be

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

IPManager version 1.6 has just been released. Below is a list of some of the features you will get on your Asterisk server using IPManager to generate your configuration files. Download: http://ipsoftware.thorben.dk <http://ipsoftware.thorben.dk/> PBX Features The following features will be available to users of the PBX if you are using IPManager to configure your PBX. *

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

Hi all. I am seeing incoming calls from digital lines (mobiles e.g.) dialling my main number + 3-digit extension just fine ("Accepting voice call from '11234567' to '250' on channel 0/1, span 1"). The problem however is with calls from analog lines: "Accepting voice call from '13331846' to '25' on channel 0/1, span 1" * just sees 2 digits, not

I have just joined the list and I have a problem, perhaps someone can help. I have: Linux 7.0 with 2.4.7 Kernel (with various upgrades for functionality). Samba is running on this machine with shares and a printer on the USB port. This machine also has a modem (demand dialling) and I am using the Monmotha Firewall script. This works quite well for the WinME PC able to access the printer and

We are trying to configure a Sangoma A101 card to allow both incoming and outgoing calls on a UK (BT) ISDN30e line with only 24 channels enabled. At present incoming calls work fine. We can't call out -- we get a BUSY/CONGESTED error. Do we need another context in our zapata.conf? In other words, do we need to reserve, say, channels 17-24 for outgoing calls? I also wonder if the signalling

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 1 or 2 channels in use out of the 16 at that time. When this happens, the user just needs to re-dial and the call goes through OK. On a SNOM phone when the problem occurs, a "Service Unavailable 907" error is shown. [2008-10-14

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx "sip notify reboot-snom mjc_home" to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. - Mike

hi asterisk users How can i make asterisk "ignore" invalid extensions, and go on playing the background soundfile? Normally, asteriks will take the user to the invalid extension if the caller presses anything other than 1 or 2 in the following context:: [example] exten => s,1,Answer() exten => s,2,Background(hello-world) exten => s,n,Goto(s,2) exten =>

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: >Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 >Apr 30 11:01:21 WARNING[12785]: acl.c:244

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there. I've done some googling around to try and find an example of what I'm trying to do, but it's one of those things that just seems hard to find the right terms to search for. If there's some documentation out there on this, I'd appreciate being pointed in the right direction. If not, then if someone has some

Hi, I have recently started a job as a system administrator, and as part of my responsibilities I have to look after an asterisk system. Quite impressed with it, but have one or two niggling issues. One of the last things my predecessor here did, was install a VoiceBlue mobile gateway unit, and though it seems to work ok, nearl 20 seconds pass from dialling a number to the call connecting, which