similar to: Looking for PRI Outbound Caller ID Configuration

As an employee in the technical operations of a CLEC this information is easily obtainable by anyone that has access to the Class 5 switch servicing that PRI... A Q.931 trace in the Class 5 Switch will tell the whole story.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Tuesday, June

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendant/voicemail dance. My problem is I can't find an application command to cancel the

David, I guess I'm a little confused here. Are you asking me to provide a "pencil copy" of an LOA for your review? I don't understand why you need an LOA from us. We need an LOA from you to order circuits that will be billed to us that will be attached to your CFA. It was also my understanding that you had an LOA ready to be given to us, which had already been reviewed by

There are often times that I want to read a DB value from the dialplan, and if this family/key pair does not exist, set it to some default value. for example: 1234,1 => Set(EMAILADDR=${DB(x/y)} 1234,2 => GotoIf($["${EMAILADDR}" = ""]?3:4) 1234,3 => Set(EMAILADDR=Someone@test.com) 1234,4 => NoOp(${EMAILADDR}) 1234,5 => Hangup() I have modified the db function

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html I saw this post about the M(x) option for the Dial command, but I could not find a reply questions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command. I've tried to add arguments to the macro by using the syntax M(x,arg1), and I always get the

I am trying to retrieve the cause code of a outgoing call over a PRI where the number called is out of service. When an out service number is called I get a recording that the number dialed is not a working number. I see cause code 1 in in the CLI as soon as the call is dialed the Telco recording goes on for 30 sec. then hangs up. Any idea on how retrieve info that the called number is is

Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable is not defined, I will get a parse error. Yeah, there are ways around it, but I would think that it should return false if 0, null, or undefined. I would change it, but I have no idea about bison and I only have very basic C skills. There was a bug opened on this, and there was a valid work-around posted,

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

hi list, I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27. I use an e1 card with sip clients. My extensions look like this: [E1] <snip>...<snip> exten => 33006733,1,Set(CALLED=${EXTEN}) exten => 33006733,2,Dial(SIP/1@192.168.0.23) exten => 33006733-ANSWER,3,Answer() [SIP] exten => _X.,1,Noop() exten =>

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti

hi all! I'm running an Asterisk-box with bristuff-....RC8n and 2 HFC-S cards. I m located in Vienna/Austria. I have the problem that on outgoing calls i hear my voice as a short echo (about half a second). This occurs not on every call. I tried some changes in my zapata.conf, with rxgain and txgain settings, but to me its hard to find a configuration which is good for every call i make. Is

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten => 1234,1,Dial(SIP/providerA) exten => 1234,2,Dial(providerB) exten => 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a

Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360?

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed