similar to: How to disable authorization during Incoming calls to asterisk

dear all, we need help for integration asterisk (sip) with mera we have configure for sip.conf and extentions.conf sip.conf [mvts] context=mvts type=friend host=10.10.0.2 dtmf=rfc2833 in extentions.conf [mvts] ; ; mvts exten => _01162.,1,SetCallerID(mvts) exten => _01162.,2,SetCIDName(to mvts) exten => _01162.,3,Dial(SIP/${EXTEN:3}@mvts) i need if i dial 01162 in mera replace with

I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT) when the asterisk box has a dynamic IP address. If the Asterisk box has a fixed IP, everything is OK. Any ideas? I'm looking for a working sample of the sip.conf in this case... user.cfg (for MVTS) is also appreciated if any special setting should be done there also.

Hello. I'm try to set up asterisk for making outgoing calls with oh323 channel driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Am 13.06.2020 um 08:28 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi there. I have the similar problem ("Digium fax - sending fax call file vs manager originate") sending faxes with Asterisk 1.6.2.6 and Digium res_fax. Receiving is OK. I use Local channel in Call file and context [fax-out] in dialplan. My setup: Asterix<-SIP (T.38)-> Cisco(MERA MSIP v.1.0.2)<-> LocalTelco<->fax machine Debian GNU/Linux 5.0 ; Linux 2.6.26-2-686

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

Hi, i'm trying to install in wine-1.0.1 on Ubuntu 9.04 a management software but I receive run-time 713 error. Any solution or tip? Here the log.txt: fixme:ole:OleLoadPictureEx (0x12c8c44,35146,1,{7bf80980-bf32-101a-8bbb-00aa00300cab},x=0,y=0,f=0,0x32fac0), partially implemented. fixme:ole:OleLoadPictureEx (0x12c8c44,774,1,{7bf80980-bf32-101a-8bbb-00aa00300cab},x=0,y=0,f=0,0x32fa90),

I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background("OH323/RXXXXX", "vm-extension") in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Greetings- I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Of course, logs on Box A were not set to show debug

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using

Hi, I would like to restart the conversation regarding adding SVE support to LLVM. This time I am framing things from the code generation point of view because our immediate priority is llvm-mc support rather than auto-vectorisation. Can you please review the following text outlining MVT changes we would like to make so SVE instructions can be added to the AArch64 Target. My overriding

On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote: > On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: > > I do not want set allowguest=yes. The problem is, there is no official > > list with ip addresses of Telekom Germany. But I think all ip > > addresses comes from the ip range 217.0.0.0/13. > > Hello Daniel, > > Judging by the lists

Hi list! Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180

Here is a series of patches replacing EVT with MVT at a number of places in TargetLowering. The last two patches are related cleanups in SelectionDAGBuilder. /Patrik Hägglund > git log --stat --reverse origin/master.. commit 8dabe3eb005360347eabb86a2e88c3b6e9098ed5 Author: Patrik Hägglund <patrik.h.hagglund at ericsson.com> Date: Tue Dec 4 10:37:37 2012 +0100 Change