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Call Cannot Be Completed As Dialed Freepbx

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Do you know which version of the firmware your unit has? CRTC Net Neutrality Debates [CanadianBroadband] by thehammer86551. Dohhh (John Martin) 2014-10-19 13:53:48 UTC #15 I did not have any dial patterns setup. Constantin 2014-04-28 18:49:16 UTC #15 Thank you all for your quick replies. weblink

dicko 2014-10-19 03:26:26 UTC #8 yet its not working. . . Any help would be appreciated. Most annoying. Can you suggest any configuration changes I could make?

Silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer

E-mail mket Top Re: SIP-to-PSTN calls completed approx. 25% of the time. cosmicwombat 2012-04-05 01:21:34 UTC #2 Did you set up outbound route(s) ? Under Applications, go to Misc.

E-mail mket Top Re: SIP-to-PSTN calls completed approx. 25% of the time. Eriond configuration worked as a charm for me.I managed to fix Teletek SIP trunk configuration in 3 minutes. Well I spent a little time fiddling with the Linksys control panel and found that doubling the value of "PSTN Dialing Delay" from 1 (second) to 2 solves the problem (or Your Call Cannot Be Completed At This Time I see QDIALER referenced a few different places.; extension 28: agent custom dialexten => 28,1,Answerexten => 28,n,NoOp( "QM: Agent Custom Dial.

Hmm, I hadn't thought of that. Grandstream Your Call Cannot Be Completed As Dialed On the new machine I have the same connection info and I get the message from the new machine "call cannot be completed as dialed" dicko 2014-10-19 03:30:50 UTC #10 sorry, Dialing ${EXTTODIAL} on queue ${OUTQUEUE} made by '${QM_LOGIN}'" )exten => 28,n,Set(QDIALER_QUEUE=${OUTQUEUE})exten => 28,n,Set(QDIALER_NUMBER=${EXTTODIAL})exten => 28,n,Set(QDIALER_AGENT=Agent/${AGENTCODE})exten => 28,n,Set(QDIALER_CHANNEL=SIP/${QDIALER_NUMBER})exten => 28,n,Set(QueueName=${QDIALER_QUEUE})exten => 28,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,);exten => 28,n,Set(CALLERID(all)="1234567890" <1234567890>) ; Uncomment and change this if you pop over to these guys It helps if I read the built-in help topics.To do what I wanted to do, I have to specify the locally-dialed number in the prefix box, and the actual 11-digit number

Guess I'll rest my eyeballs and prepare for another battle. Email Reset Password Cancel Need to recover your Spiceworks IT Desktop password? Every phone company starts with Were sorry, but what are they sorry for ? :> "We're sorry, your call cannot be completed as dialed. I have just changed the provider of the SIP trunk and now I cant get outbound calls to work, I get "your call can't be completed as dialed".

Grandstream Your Call Cannot Be Completed As Dialed

I mentioned this earlier in this thread. https://community.spiceworks.com/topic/418223-freepbx-cant-make-call-without-using Hope someone can help me with this as im banging my head against a brick wall trying to work out how this is done lol. Silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer Before I had access to Eriond configuration example I had lost 1 entire day without any success. Your Call Cannot Be Completed As Dialed Meaning Fast forward to today, I get the dreaded "Your call cannot be completed as dialed" from my trusty PBX.At the time I implemented it I had Phone Power VoIP, and since

Inbound is working fine and so does extension to extension. http://cmptp.com/call-cannot/call-cannot-be-completed-as-dialed-0106.html But from a technical perspective, a dialplan is a dialplan and it should be carrier-agnostic. The last release that I saw was from Jan 2009 at version 5.1.10. -Mazzic mazzic Oldsterisk Posts: 277Joined: Wed May 05, 2010 8:55 amLocation: Waco, TX E-mail mazzic Top Re: by mket » Mon May 17, 2010 11:56 am mazzic wrote:What is the complete message for the we are sorry message? Freepbx Outbound Route Dial Patterns

Here is the tail of the Asterisk's log when I try to make a call from Twinkle: [2012-04-04 12:07:24] VERBOSE[-1] chan_sip.c: -- Unregistered SIP '101'[2012-04-04 12:10:44] VERBOSE[-1] chan_sip.c: -- Registered SIP One was my new VOIP phone number and the other was just a forward slash (/). Teletek from Sweden. check over here by navaismo » Fri May 14, 2010 2:46 pm I see, your configurations are the same because nobody inform about this issue?

Please login or register.Did you miss your activation email? 1 Hour 1 Day 1 Week 1 Month Forever Login with username, password and session length News: Welcome to the new I'll try messing around with the delays in the Linksys config.Do you know which version of the firmware your unit has? I know how to record an annoucment.

CSeq: 19 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.0 stamp 58832 Authorization: Digest username="101",realm="asterisk",nonce="76a7e4d8",uri="sip:[email protected]",response="7b612db84bcef48a394e454893379f1d",algorithm=MD5 Event: message-summary Content-Length: 0 <------------->

Server OS recommendation [Networking] by Frank_IT185. Thank you again, I am most grateful. For example to dial 9 to get an outside line you could use 9|.  would send everything after the 9 out that route. 1 Serrano OP Berrick Dec I would like to be able to stop all extentsions from being able to dial cell/mobile and premiume rate nubers.

I have not registered my SIP line for emergency calls as another route handles that, but you should probably allow for 999 as well. see ya take care. I really don't know where to go from here. this content system (system) 2014-06-04 20:10:39 UTC #18 Home Categories FAQ/Guidelines Terms of Service Privacy Policy Powered by Discourse, best viewed with JavaScript enabled Log In No Outbound -> Cannot be completed as

After deleting the forward slash object I was able to make inbound calls. Can you send me any information about your setup (sip trunk etc).Br, Constantin Eriond 2014-04-28 15:52:55 UTC #11 Right, there are a few things not documented well about Teletek. Applications and create a new application, making the "feature code" the shortcut number you want to use (note it does not have to begin with a *) and the destination the I would like to understand why I can't seem to drop the * (is this a default behavior?)for starters.

Then, I also have to tell Asterisk to manipulate the number before it hits the trunk so that 450 really means 17185551212.And well.....