similar to: sequential dialing preferences

exten => s,1,Answer() exten => s,n,Background(astcc-please-enter-your) exten => s,n,Background(zip-code) exten => s,n,WaitExten(5) exten => s,n,Read(NUMBER,,5) exten => s,n,SayDigits(${NUMBER}) exten => 22042,n,Dial(SIP/sipvendor/11111111111,120,A(ginger3)) exten => 22601,n,Dial(SIP/sipvendor/11111111111,120,A(ginger3)) ; xxxxx/ winchester exten =>

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,

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=SeptemberWLHalo3_MSNHMTxt_1

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)

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

the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egress leg only accepts g729. If this is design intent I'm wondering if there is demand enough to justify a feature request? Any advice on how I can work around this issue?

I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical

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

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

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

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

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1111 1112 1113. What i want to know

Hi all, I am configuring asterisk as a prepaid calling card. I am getting different local rate from my ISDN provider e.g 0.002 for landline and 0.13 for mobile etc. In this case I thing I have to say my asterisk/a2billing to bill based on local access number. so How can I retrieve called number (eg. 03-6832-1040 and 0120-272-060 is our ISDN PRI access number) to my asterisk server so i can

Hi, i need to match a number with like 03012345678*0 or 03012345*9 I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring that *X is required at the end. The interesting part is that like expected _X*X is matching only numbers like 1*1 and not 11 Regards Robert Wagner -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type:

It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER="Alert-Info: Ring Answer" (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do

Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration

Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full:

Hi, I'm trying to install voicemail-imap support but there seems to be a missing module: imap_tk checking for mandatory modules: IMAP_TK... fail configure: *** configure: *** The IMAP_TK installation appears to be missing or broken. configure: *** Either correct the installation, or run configure configure: *** including --without-imap. My configuration Ubuntu 14.04 LTS Asterisk