similar to: VoIP Incoming Issue

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I''m trying to get lossless VoIP traffic over my 3000k/500k ADSL line. Shaping outgoing traffic is no problem: I set total ceil for outgiong device (ppp0) to 450kbit and put VoIP into highest prio class. Even during full upload the voice is clean on the other end. Now I tried to get the same result for incoming data. I attached HTB to eth1 where the incoming voip traffic is

I'm setting up a demo for two asterisk machines. One will be a central Asterisk server which will handle everything already in VoIP (office-like functions plus agents functionality). The second Asterisk box will be used strictly as a VoIP gateway to the first server. The gateway server will have 4 T1s connected to it and what I was thinking on doing was the following: in

I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone

Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1

I am looking for a provider that will provide an equivalent of DID/DOD trunks via IAX, IAX2 or SIP using numbers in Metro Chicago (prefer Skokie), LA (prefer West Hollywood or Venice), and/or Orlando (prefer Winter Garden). If I can migrate some of my existing numbers using LNP, that would be even better, but it is not a requirement. While I know that there are several companies that will

Look at the thread Optimum online-upload throttling confirmed. It seems like throttling is done by all Cable companies and that might affect the VoIP performance, specially for uploading. Try when the activity in the cable line is low (i.e. late) and see if it gets better, or try sending all the data through your DSL line. Carlos > > Message: 26 > Date: Fri, 19 Aug 2005 15:58:59 -0400

I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is

Hi All, I read some discussions about the new SIP authentication methods for 1.6.X branches and possible addition of new type of user, type=trunk. I'm wondering about the disposition about this. Will it be added? In 1.2 and 1.4 branch, a SIP invite was first checked for a valid [user] then a valid host=ip, then if not present send call to [general] context=incoming. In 1.6, a SIP invite

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

I don't use digitalvoice, but based on a similar provider you may need to have your username inserted in your extensions.conf context.... [incoming_calls] exten => username,1,Answer( ) exten => username,2,Playback(demo-echotest) exten => username,3,Hangup( ) Just an idea....

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

We are setting up asterisk with Teliax and having trouble getting the incoming call to work all the time, the outgoing does not seem to have a problem. I have worked with their support but since they say that we are getting the initial call to our server they want to charge to take a look. They did a tcpdump and we are seeing an attempt but no CLI most of the time. Some times we see this but it

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ... interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table

Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone> I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in. Previously,

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

Trying to get some straight info from the VOIP providers is difficult. Say there's a small Asterisk switch and it's registered with Broadvoice or LiveVOIP or someone. There are a couple of people using the switch, one is on an outgoing call with the VOIP provider. What happens when someone else initiates another outgoing call through that provider on the same SIP registry? Does * know

I have been looking at chan_bluetooth, so far being unable to make it compile with Asterisk SVN trunk. I was wondering about the different ways it can be used. What I have read so far implies two possibilities: 1. Asterisk pretends to be a handsfree unit, and can use the cell phone for placing calls over the mobile network, or answer inbound calls from the mobile network. 2. Asterisk

I need to accept an incoming call, make a series of outgoing calls, and once I find someone willing to accept the call, bridge the original incoming call to the outgoing call. Using Dial from an AGI script isn't enough because once the Dial'ed number connects, the call is immediately bridged and I need to ask the called party if they will accept the call. I can see a couple of