similar to: Set origin CallerID when forwarding calls to mobile phone

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi all, I've got 4 actors on my stage: Alice calling from outside Bob transferring incoming calls to Charlie Charlie who has a mobile phone My PBX which is connected to my ISDN line. I want Charlie to see Alice's Callerid after Bob has transferred the call as if Charlie is receiving the call from Alice, transparently. Tried to set the callerid but Charlie sees my telco line number,

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have any "normal" telephones connected). What i wanne do is somethink like this: Subject: +49691234567 Attache: *.pdf The attched pdf have to be send ;) -- Mit freundlichen Gr??en Daniel mailto:daniel at listmails.de

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

Hello, does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our callerID is no more seen by them. We use Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or equal to CALLERID(num). We tried

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

Hi everybody, I would like to know if it is possible to add a suffixe number to my official sipgate number? For example my official sipgate number is 123456 My local users numbers are 0001 0002 Could it be possible to compose from internet 1234560001 or 1234560002 , then the message could be forwarded to user 0001 or 0002. Actually I can only forward my sipgate number to only one user. So I am

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Regards Mattias -- Mattias Andersson -------------------------------- Storskiftesv?gen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38

Hey Guys, I have something that just started happening. When my users call each other on their 5 digit extensions their CallerID is showing as XXXXX at 10.25.2.50 (XXXXX would be their Ext. and 10.25.2.50 is my server) Calls in an out to the outside world are fine. I have scoured my configuration and can't find what would be causing it. I have checked the sip.cfg in the polycom's and

Hi, I have researched more into the problem of my Asterisk set-up not answering calls. The following error was shown on the CLI, can anyone explain what the problem causing Asterisk to not answer the SIP calls be? Information: I have an Asterisk box on a home LAN, behind a D-Link router/firewall connected to a cable modem. The 82.x.x.x is the IP for my cable modem. 192.168.0.101 is my

On Thursday 18 June 2020 at 19:57:03, Administrator wrote: > does some people here use https://voicetrading.com which is a Dellmont > service from Netherlands. At the high begining they were also known as > Finarea (CH and DE mixed Co) > Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or > equal to CALLERID(num). We tried replacing + with 00, same problem. >

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222

I can't see for the trees :) I can make calls out to my SIP provider but get an "unable to authenticate <calling no.> when I try to call in via the PSTN number they have supplied me (where <calling no.> = phone number trying to make the call) sip.conf [general] register => 4316568:xxxxxxxx@sipgate.de [sipgate] secret=xxxxxx username=4316568 fromuser=4316568

I am a new trixbox user. One of the things I'd like to get working is being able to tell if a user is calling me by directly dialing my extension, or if they pressed 1 for sales. When they press 1, it rings a group of phones, and the call is almost always for someone else. So I'm always looking at my phone when it rings, trying to recognize the incoming number and decide if I

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com