Re: [asterisk-users] rtptimeout: how to detect it in dialplan?

On 18 Jan 2013 15:22, "Klaus Darilion" wrote: > > Hi! > > I want to forward a call to another destination if the outgoing call leg has an rtptimeout. But as far as I see there is no way to find out if the hangup was due to a rtp timeout or any other reason. I thought that HANGUPCAUSE or DIALSTATUS

[asterisk-users] Bridged Digital call

Hi All Just upgraded from 1.6? to 1.8.4.1 I ised to be able to get a digital call working across a bridged isdn channel in 1.6 and 1.4 using the following;- exten => _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:) exten => _X.,2,dial(DAHDI/g1/${EXTEN}) exten => _X.,3,Noop(${CHANNEL}) exten =

[asterisk-users] CDRs in 1.8

I'm using ISDN30 for a bridged application in all the old versions of asterisk the time slot number is shown in the channels and dstchannel fields of the cdr I understand this has chaned in 1.8,is there a way of getting the time slot information stored somewhere at the end of the call so this can

Re: [asterisk-users] Bridged Digital call

both show transfercapability DIGITAL Regards Robb On 16 June 2011 23:40, Richard Mudgett wrote: > > Hi All > > > > Just upgraded from 1.6? to 1.8.4.1 > > > > > > I ised to be able to get a digital call working across a bridged isdn > > channel in 1.6 and 1.4 using the following;- > > > > > > ex

Re: [asterisk-users] Bridged Digital call

any reason why this would happen, should I report a bug on the issue tracker? Robb On 17 June 2011 19:55, Richard Mudgett wrote: > > > > Hi All > > > > > > > > Just upgraded from 1.6? to 1.8.4.1 > > > > > > > > > > > > I ised to be able to get a digital call working across a bridged > > > > isd

[asterisk-users] CDR dialed digits missing

Hi I'm using asterisk 1.6.2.18.1 I'm having a problem where only the first four digits are collected in the cdr when the call is dialed overlap but if the call is dialed en-block the whole dialed digits are recorded chan_dahdi.conf [trunkgroups] [channels] language=uk switchtype=euroisdn pridi

[asterisk-users] Home line noise problem

I Have a home line connected to a tdm400p with 3 extensions and a siemens sip-dect , it seems to work fine but during a call there is always a digital squeal every so often does anyone know what this could be? Robb ___ -- Bandwidth and Colocation Provide

[asterisk-users] Meetme

Hi All I would Like to run a macro in a meetme conference when a user presses a certain digit sequence, but I cannot seem to find how to do this , is it possible? if so how? Thanks for you help Robb ___ -- Bandwidth and Colocation Provided by http://ww

[asterisk-users] TDM400P alarm state

I'm having real problems with my connection to BT, it is a home line, but after a while it sets an alarm and only a restart of asterisk resets it could some one look at the below configs and suggest any changes to make this more reliable Thanks for your help Robb asterisk version 1.6.1.10 dah

[asterisk-users] Linksys SPA922

Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows "Anwsering" but never does and the far end continues ringing until the voicemail answers, this then show as a disc

Re: [asterisk-users] Linksys SPA922

/7 Danny Nicholas > Show us your CLI output. I suspect that you’re not getting a bridge > and/or you’re timing out. Also sip.conf and user.conf would be helpful as > well as Asterisk release. > > > -- > > *From:* asterisk-users-boun...@lis

Re: [asterisk-users] Digium PRI cards for data usage?

Do you have to set aside kines for the data channels or can you have dynamic data channels, for example ISDN dialup internet backup? Robb 2009/9/1 Tim Nelson > - "Tilghman Lesher" wrote: > > On Monday 31 August 2009 21:59:28 Tim Nelson wrote: > > > Greetings- I'm wondering if the Digium P

[asterisk-users] portech MV-378 SIP GSM Gateway

Hi All I having an intermittent problem with the above mobile gateway and would appriciate some advice basically 1 in 10 calls fail at some point during the call, the duration of the calls ate completely different call progression Call comes in from Zap channel and dials a mobile number on the

Re: [asterisk-users] Avaya 9640 Convert to SIP (slightly off topic)

its on the Avaya site, but basically, you need to have the sip image and the 96xxsettings.txt and the 96xxUpgrade.txt on the route of a web server, the sertup the phone to read its files from the web server, normally this is by pressing then a-d-d-r then # set the ipaddress, callsvr and subnet a

[asterisk-users] PrivacyManager

Hi thwe PrivacyManger app states thast you can use a context to match against for the input , but gives no real examples or explaination, does anyone have a an example context for this Thanks in advance Robb -- _ -- Bandwidth a

[asterisk-users] TDM 400p and Noise on the line

Hi I wonder if anyone has any sugestions I have had a TDM400 for a couple of years, and I have always had problems with noise on the line, so tonight I have been doing some research and have found that if I load the CPU dahdi_test has almost perfect results and no noise dahdi_test Opened pseud

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