similar to: This should be easy: What happens when the Calling Party hangs up

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong. One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMITEL123456 1296197222 at entryhome<mailto:1296197222 at entryhome> Ringing

All, Is there a way I can perform a lookup to see if a given extension exists within a given context, on the local system? I could call Dial() and check the result of $DIALRESULT, but I'm thinking there should be a better way. Note, that I don't want to use ChanIsAvail(). That's only for determining if endpoints, ie phones, are registered. It doesn't seem to work with extensions.

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

Is it possible to have both a busy and an away message when the call waiting feature is enabled? extensions.conf ... exten=>403,1,Dial,Zap/3|10 exten=>403,2,Voicemail2,u403 exten=>403,103,Voicemail2,b403 ... Because I have enabled call waiting, I can't see how it will be possible to get the busy message to play (because there will always be a dial tone). Am I right, or do I have

I'm still struggling to pickup calls. I now have a single context (entryocginternal) where I have "include => parkedcalls". The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

Hi BACKGROUND INFO:I have a new Eaton 5E 1100i USB UPS which their website claims is supported on Linux. They don't provide software for it for Linux, only for Windows.Referring links: https://www.eaton.com/za/en-gb/catalog/backup-power-ups-surge-it-power-distribution/Eaton_5E_UPS.html https://www.eaton.com/za/en-gb/skuPage.5E1100IUSB.specifications.html It was manufactured in May 2020

The Asterisk Development Team has announced the release of Asterisk 1.4.38. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.38 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.38. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.38 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a list of issues resolved in this release: * Fix

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a list of issues resolved in this release: * Fix

On 2020-08-19 17:36, Louis van Dyk wrote: > Hi > > BACKGROUND INFO: > I have a new Eaton 5E 1100i USB UPS which their website claims is supported on Linux.  They don't provide software for it for Linux, only for Windows. > Referring links: > https://www.eaton.com/za/en-gb/catalog/backup-power-ups-surge-it-power-distribution/Eaton_5E_UPS.html >

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I don't want building1 to access international, but

Hi All, I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2 trunk between the 2 servers using the guide on dumbme. Trunk is not using register string and no authentication. In my dial plan I have 7XX numers on server B and 6xx numbers on Server A. Calls from my SNOM phones are ok between the extensions on the 2 servers. In server A I have a eicon 4 port BRI card connected.

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I done't want building1 to access international, but does

The Asterisk Development Team has announced the release of Asterisk 12.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 12.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile