similar to: FW: [asterisk-dev] Call failed, reason 0 explanation.

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813.

> > >On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote: > > >>>> >Freddi Hansen wrote: >>> >>> >>>>> >> with boards from Aculab, we are replacing Aculab boards with Digium >>>>> >> boards BUT we would need more >>>>> >> Digium boards IF we could use both Digium and Aculab cards in

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Variable Name needed That wasn't it either. I tried a few other likely fields from

hi below is sysvinit script for handling tcng-script .... please excuse my bad bash skils, correct me where i''m doing stupid things :") what is left... many things, if i have time i will implement them too : - start [devices] - and then correct handling of service lock files i.e. per device lock file - stats blah ... - all ideas are welcome !! - test or some such, i mean the

Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the

I checked https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information But I cannot find a way to insert CALLERID(ani2), which I can read, but when I try to set it for a new call, I get a runtime error. This information, known as isup-oli comes embedded in the From header,like this <sip:9087311878 at 64.45.157.104:5060;isup-oli=62>;tag=sansay1724414rdb124 and it can be read

Hello list, I'd like to know if there is a function that tells in which package is a given function. Something like : which.package("lda") [1] "MASS" Thank you. Romain -- ~~~~~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ~~~~~~~~ ~~~~~~ Romain FRANCOIS - http://addictedtor.free.fr ~~~~~~ ~~~~ Etudiant ISUP - CS3 - Industrie et

Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did

Hello, I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. I am able to make calls outbound through the gateway, but I am not able to make calls into the PBX from external PSTN. Specifically, an incoming call is _received_ by Asterisk, but it is not able to route the call internally owing to the following error: [Feb 18 21:08:47] NOTICE[4606]:

Hello list (happy new yeaR), Here's a copy of a message i just send to Christian Hennig (who wrote the fpc package). That may interrest some of you, and maybe someone could have a better solution than mine. Romain. ------------------------------------------------------------------------------------------ Mister Hennig, [[[ I'm writing in english because i don't know german

Hello, Just look at the examples in ?persp. There is a function called trans3d defined in it that will traduce your 3D coordinates to 2D, and so you will be able to draw lines with lines function. Romain. Corey Bradshaw a ??crit : >I've created a perspective plot using 'persp' in the graphics package. >I'd like to add a second plane of z values to the existing plot, but I

Dear wizaRds, I use the MASS::kde2d function to estimate density of the two first principal components. I do that to have a graphic visualisation of a "group structure" in my dataset. So far, no problem. But i would like to estimate that density using weights according to the COS?? values that tells me if my observation is well represented on the factorial plan 1-2. I would like to

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

Hello useRs and helpRs, Some time ago, in a gallaxy far away (here is the thread : http://finzi.psych.upenn.edu/R/Rhelp02a/archive/46532.html ) we discussed about a graph gallery showing the power of R in that domain. I did some work around that, and there is a (pretty advanced) draft here : http://addictedtor.free.fr/graphiques/displayGallery.php For instance, there are some of my graphs,

Hello, Didn't get much help on asterisk-dev, so here it is. Please help. Thanks. From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mark Hamilton Sent: March 13, 2008 2:10 PM To: asterisk-dev at lists.digium.com Subject: [asterisk-dev] Hardware and CentOS tweaks. Hello, We're working on using Asterisk as an

Hello, There are some parameters in the ISUP data (coming into the network via SIP-T packets) that need to be translated into SIP headers to be used by asterisk for proper call routing. What gateways are available to accomplish this? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is SS7 and PRI in any way compatible in that if the interface is configured one it will work for the other (granted, it will not have any of the ISUP, etc. parameters available if the line is PRI) or are they two distince protocols that have incompatible signalling? Thanks, Elliot