similar to: H323 dial problem

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is!

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when I

I am installing R_3.5.1 from source on ubuntu 18.04, and 'config' + 'make' gives me (at the end) Loading required package: Matrix Error: package or namespace load failed for ?Matrix? in dyn.load(file, DLLpath = DLLpath, ...): unable to load shared object '/home/goran/src/R-3.5.1/library/Matrix/libs/Matrix.so': /home/goran/src/R-3.5.1/library/Matrix/libs/Matrix.so:

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

Dirk, thanks for your help. At work I have (ubuntu 16.04): ii libblas-common 3.6.0-2ubuntu2 amd64 Dependency package for all BLAS implementations ii libblas-dev 3.6.0-2ubuntu2 amd64 Basic Linear Algebra Subroutines 3, static library ii libblas3 3.6.0-2ubuntu2 amd64 Basic Linear Algebra Reference implementations, shared library and everything works before

Hi there, On 1 December 2017 at 23:24, G?ran Brostr?m wrote: | Dirk, | | thanks for your help. At work I have (ubuntu 16.04): | | ii libblas-common 3.6.0-2ubuntu2 amd64 Dependency package for | all BLAS implementations | ii libblas-dev 3.6.0-2ubuntu2 amd64 Basic Linear Algebra | Subroutines 3, static library | ii libblas3 3.6.0-2ubuntu2 amd64 Basic Linear

I don't think that will fix my problem. The hints on the individual user extensions (101, 102, 103 and 104 below) are working just fine. sccp.conf example of 1 user: [devices] type = 7970 description = User1 tzoffset = -6 autologin = 101,401 speeddial = 102,User2,102@wct-internal speeddial = 103,User3,103@wct-internal speeddial = 104,User4,104@wct-internal device => SEP000F90CEF9D3

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have 4 employees and we're running Cisco 7970 phones. Each phone has a unique SCCP line configured (in the autologin area of the sccp.conf file) for each employee. We have hints set up in the extension.conf file like the following: exten => 101,hint,SCCP/101 exten => 102,hint,SCCP/102 exten => 103,hint,SCCP/103 exten => 104,hint,SCCP/104 We have speeddial= lines set

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

On Tue, 9 May 2006, Nick Lewycky wrote: >>> For an analysis pass, I'm interested in reading the lattice values >>> calculated by SCCP. Should I split the current SCCP optimization into an >>> analysis piece and the optimization that depends on it, so that I can >>> use its analysis results? >> >> SCCP is already split into an SCCPSolver class

I am trying to register cisco 7920 to asterisk using sccp since to sip firmware upgrade to it ,but its ends with failed registration.Can you please send me a sample for sccp.conf configuring cisco 7902. Thanks -- SCCP: Accepted connection from 192.168.5.163 -- SCCP: Using ip 192.168.5.228 -- SCCP: Accepted connection from 192.168.5.163 -- SCCP: Using ip 192.168.5.228

Chris Lattner wrote: > On Tue, 9 May 2006, Nick Lewycky wrote: > >>>> For an analysis pass, I'm interested in reading the lattice values >>>> calculated by SCCP. Should I split the current SCCP optimization >>>> into an >>>> analysis piece and the optimization that depends on it, so that I can >>>> use its analysis results?

Hi, This is a message I already posted on the chan_sccp mailing list, but since this list has a lot of active members, I'm hoping someone might be able to help (And my problem is * related, so I guess it's ok if I post it here also ;) ). I'm trying to get SCCP ATA188s to run with Asterisk. The Asterisk box uses the latest Asterisk@Home image (Version 2.6). I have compiled and