similar to: Caller ID sip trunk

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

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

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

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 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" ,

As you my be aware the X100p cannot collect uk caller id, now I have a modem and a perl script that creates a file /etc/asterisk/callerid.txt on each incoming call in the format number,date,time,name over writing each time a new call comes in I can asterisk read this file and populate the callerid for internal phones and cdr? I think it can be done with AGI but don't know where to start

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 everyone, I'm in the very early stages of rolling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any

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

Hi list, I'm hoping that I'm being stupid, and someone can tell me what's going on, but for the life of me I can't figure it out. (it's been a long day, and I'm now in the last 3 weeks of organising my wedding, so I hope this makes sense ;) ) When at my desk, accessing (for example) my voicemail, the dtmf tones are passed perfectly, I can enter password, change

What's going on here? 555* seems to indicate that the number is being passed as the callerID because NoOp says the phone number. I'm trying to emulate cell phone voicemail where you call your own number to check your voicemail. -- Accepting AUTHENTICATED call from 65.182.165.XXX: > requested format = gsm, > requested prefs = (), > actual format

Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

Hello, I've got an Asterisk system with 3 SIP trunks configured. Each SIP trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, all set with max channels to 4. Unfortunately, when the first trunk reports a "480 Service Unavailable" (all ports in use), Asterisk reports congestion without

I have a question for those of you out there using VoicePulseConnect for incoming did I have in my Realtime extensions Database (the x's are replaced with my phone number) context = voicepulse-in-01 exten = xxxxxxxxxx/ Priority=1 app=NoOp appdata = Incoming call with no callerid on xxxxxxxxxx However it never triggers I also tried using one of my other providers (voipjet for outbound) and

I've some code to manipulate incoming Caller-ID - so its suitable for replying to... [sipdef] exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>) ; Alter incoming calles from pulver - add a '87' exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4) exten => s,3,SetCIDName(87${CALLERIDNUM}) exten => s,4,SetCIDNum(87${CALLERIDNUM}) exten

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

Hi all I use sipgate and FWD but seem not to get it going. I do not have NAT on the asterisk box (static ip). The asterisk box has 2 network interfaces. One internal and one external. Now when I make an call to a FWD or SipGate number all I get is -- Executing NoOp("SIP/113-6d2e", "") in new stack -- Executing Goto("SIP/113-6d2e",

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

hello i had configured a Dial plan in which i am using application time base i.e 1 - if? call comes to PSTN line??? from 8pm? evening till morning 8am the call should? automatically forward to guard exten i.e exten 211, and if guard dosent receive call in 30 secs message should be saved in voicemail. 2 - if call comes in working hours than it should be received by ext 112 n from there using

Hi I have installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context | exten | priority | app | appdata | description