search for: callwithus

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWL...

Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2) exten => s,n,Hangup() my sip.conf file [general] context=default allowoverlap=no bindport=5060 port=5060 bindaddr=0....

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number, I do not see any traff...

...On Tue, Aug 2, 2011 at 3:51 PM, Ryan McGuire <rdmcguire01 at gmail.com> wrote: > Running build 1.8.5.0 (compiled from source) I seem to be having an issue > with codec negotiation. I have a Grandstream HT503 FXO port connected to a > pstn line, a Polycom SP501, and a SIP trunk with callwithus. > > What I'm essentially looking to accomplish is for ulaw or g729 (preferably > ulaw) to be used to the Grandstream FXO or any other internal endpoint, and > for g729 only to be used outbound to my SIP trunk. > > Here are the basics of my config, showing the codec list from...

...machine with a similar configuration. Thoughts? I know I posted this yesterday but was hoping for some more creative comments! Zip*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip.callwithus.com:5060 N xxxx 105 Registered Tue, 07 Dec 2010 02:36:43 1 SIP registrations. my sip.conf [general] context=default allowoverlap=no ;bindport=5060 port=5060 bindaddr=0.0.0.0 canreinvite=no ;if your asterisk box is behind a...

...Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two numbers Here is what I have now which works fine for the one and only number........... exten => s,n,Dial(SIP/callwithus/12135551212,120,A(ginger3)) ; Service line so, will this work ........... ???? .......... exten => s,n,Dial(SIP/callwithus/12135551212[&SIP/callwithus/12145551212],120,A(ginger3)) ; Service line Please send comments to make this work. Thanks

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

...> actual format = alaw, > host prefs = (alaw), > priority = mine -- Executing [0033661681 at fax-out:1] NoOp("IAX2/iaxmodem-1", "we are at fax-out") in new stack -- Executing [0033661681 at fax-out:2] Dial("IAX2/iaxmodem-1", "SIP/callwithus/0033661681") in new stack -- Called callwithus/008675533661681 -- Starting simple switch on 'Zap/1-1' -- SIP/callwithus-082370a8 is making progress passing it to IAX2/iaxmodem-1 [Mar 28 01:54:56] NOTICE[16754]: chan_iax2.c:6025 update_registry: Restricting registration for...

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound to my SIP trunk. Here are the basics of my config, showing the codec list from "sip show peer <peer...

VoipRaider the site, says calls to landlines in Brazil is FREE within the freedays period. Log in to the website and hire the service, it says that I have 90 days of freedays paying for cheaper service is $ 10.. That is from what I understand, I will pay 10 dolares for unlimited call in landlines for a period of 90 days? Is that it? Has anyone tested it there? How many simultaneously calls can

What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinuing support for IAX2 I was with provider "callwithus" but they discontinue IAX2 I switched to "checkbox.cc" but they discontinued it as well. What is wrong with IAX2? -- Joseph

...st if it's an answer to something on list unless specifically asked to do so. On Sat, Jun 26, 2010 at 10:43 PM, Thomas Perron <thomas.perron at gmail.com> wrote: > It kinda did not work. > > exten => s,n,Set(CALLERID(name)=label${CALLERID(name)}) > exten => s,n,Dial(SIP/callwithus/12025551212,120,A,(demo-thanks)) > > I am dialing a DID number 1 703 444 5555 from 202 555 1212 > I see that the call enters the dp properly and then dials the 202 number > however, i do not see any change. i see that the call is coming from > 202 555 1212 > i want to see that it...

I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

If you asked this question on the biz list you would get a lot of people that will tell you that they offer services where you can set the caller ID to what ever you want. To name a few:: Nufone Teliax Voipjet ----- Original Message ----- From: "Doug Crompton" <doug@crompton.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex