similar to: Incoming SIP line does not display CallerID correctly

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

I can't seem to get consistant outbound caller ID working correctly. I have set the fromuser and callerid field in my sip.conf for my TelaSIP peer, but half the time it shows up as "No Caller ID" on my cell phone, other times it shows it correctly. Using asterisk CVS. Any ideas? Doug

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/7e1e68d1/attachment.htm

I wish to have my Grandstream GXP-2000 phones make a different distinctive ring for internal calls ( Internal ) or if the incoming call has no caller id 'NOCID'. The Grandstream phones calls allow 3 distinctive rings depending on the caller id. I have one set up and working for 'Internal' calls but unfortunately the same tone will ring if caller id is absent on a call. My

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

I've had the same excellent responsiveness from telasip, on the rare occasion that I've had issues. YMMV -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Vile Sent: Thursday, April 06, 2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Telasip

I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one

II noticed that Caller ID presentation is not making it to my cell phone through outound Telasip calls and I don't know why. It may very well have been this way for awhile (or always, not sure I called my cell phone during telasip testing). Does Telasip expect a different format than SetCIDNum(NXXNXXXXXX) ? It has always worked for the Teliax lines. BUT--- It doesn't have a problem

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this

Hi, I'm seeking to change my service provider (after ten months, I've had it with broadvoice), but I would like to keep my 310 number. I've been digging through the lists of other providers and am considering telasip (good plans and support number transfers). My concern is what precisely happens when a number is transferred from one service provider to another. After the transfer is

My DID with Telasip is disconnected and my Asterisk box won't register with them. Anyone else having problems with them? Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050814/886ceb38/attachment.htm

Am new to Asterisk - have it up and running & connected to a couple service providers (telasip & teliax). Nice! Am running our Asterisk PBX on a static DSL connection (~600 kbps up/~4mbps down), and would like to extend VoIP service to 10 non-profits we're working with. Am I correct in assuming that all calls from each organization would route through our Asterisk server & be

Hi all, I have configured the queue below, but when I go into the queue, asterisk does not announce hold time: [support] musiconhold=>default strategy=ringall context=check_time timeout=20 wrapuptime=1 maxlen=3 announce-frequency=5 announce-holdtime=yes joinempty=no leavewhenempty=yes reportholdtime=yes I've tried changing timeout, announce-frequency, but still the same; queue works

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

Hello, I define SIP registrations as follow in sip.conf : register => number:passwd at sip-server example : register => 3333333333:mypass at ip_sip_server But apparently the SIP 'contact' header in the SIP REGISTER looks like this : /Contact: <sip:s at ip_my_asterisk>/ How come ? And how to change this so it reads : /Contact: <sip:/3333333333/@ip_my_asterisk>/