search for: telasip

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...

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 insecu...

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

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 HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070224/a6865d0f/attachment-0001.htm

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension...

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...

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

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 pro...

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

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 customers can call them as a local call AND they can dial out UK numbers as local calls. The obvious side benefit would be...

...call-terminated) exten => s,7,Goto(aa_chris_start,s,1) exten => s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS DIALED, ITS LOCAL, PREPEND THE AREA CODE exten => s,9,SetCIDNum(9999999999) exten => s,10,Dial(${IPTRUNK}/360${DIALNUM},,T) exten => s,11,Dial(SIP/360${DIALNUM}@telasip-gw,,T) exten => s,12,Playback(all-circuits-busy-now) exten => s,13,Goto(aa_chris_start,s,1) exten => s,14,SetCIDNum(9999999999) ;NUMBER ISNT 7, OR LESS THAN 5 SO AREA CODE WAS ADDED exten => s,15,Dial(${IPTRUNK}/${DIALNUM},,T) exten => s,16,Dial(SIP/${DIALNUM}@telasip-gw,,T) exten =&...

...call-terminated) exten => s,7,Goto(aa_chris_start,s,1) exten => s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS DIALED, ITS LOCAL, PREPEND THE AREA CODE exten => s,9,SetCIDNum(9999999999) exten => s,10,Dial(${IPTRUNK}/360${DIALNUM},,T) exten => s,11,Dial(SIP/360${DIALNUM}@telasip-gw,,T) exten => s,12,Playback(all-circuits-busy-now) exten => s,13,Goto(aa_chris_start,s,1) exten => s,14,SetCIDNum(9999999999) ;NUMBER ISNT 7, OR LESS THAN 5 SO AREA CODE WAS ADDED exten => s,15,Dial(${IPTRUNK}/${DIALNUM},,T) exten => s,16,Dial(SIP/${DIALNUM}@telasip-gw,,T) exten =&...

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 complete, when someone dials my number, will it go to broadvoice's servers/routers initially, and get bounced over to telasip...

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...

...unately the same tone will ring if caller id is absent on a call. My solution is to insert a caller id number of 'NOCID' if there is no caller id to have separate ring tones for 'NOCID' and "Internal' calls. I have gotten this far for the nth line in my extensions.conf [telasip-in] context but need help with the syntax. In Asteriskish it would look something like: exten => s,n,GotoIf( NO ${CALLERID} then SetCIDNum(NOCID) I really wish to be able to pick up an Internal call without thought but don't really like getting NOCID sales and other annoying calls. Note...

...nge, if you're using AAH. $strUser = "admin"; #specify the password for the above user $strSecret = "amp111"; #specify the channel (extension) you want to receive the call requests with #e.g. SIP/XXX, IAX2/XXXX, ZAP/XXXX, etc $strChannel = "Local/15555555555@outrt-001-telasip"; #specify the context to make the outgoing call from. By default, AAH uses from-internal #Using from-internal will make you outgoing dialing rules apply $strContext = "from-internal"; #specify the amount of time you want to try calling the specified channel before hangin up $strW...

...om the [410] extension specified. In the [grandstream] template example below, I'd like to specify in the template that it should use the udername=410. This is the template: [grandstream](!) ; template for Grandstream sip phones ;=============== context=default ;Dials out to telasip-gw for grandstream type=friend qualify=yes insecure=very host=dynamic canreinvite=no nat=no ;add these lines to the template: username=${extension I am now in} callerid=${extension I am now in} This is one of several extension used presently: [410](grandstream) ; Karen Office username=410...

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host proxy.ideasip.com, port 5060

Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is

...'s dependent upon the specific media gateway that terminates the call, and your ITSP's rate-deck may push you opportunistically to different media gateways. Yesterday my sip-terminated Polycom IP-650 behaved exactly like the Norstar systems described above, today it is perfectly reliable (Telasip.com). TRY IT FOR YOURSELF (dial-in numbers below): For BOTH routines (one phone number for each), press a digit, Allison will say it back to you: As of this posting, RelaxDTMF is OFF. I will leave this configured for at least 48 hours. TEST -- WaitExten() Call 312-445-5905 to run the [without-d...