similar to: inbound call issue...

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status:

Hi list, I have for a year I have an account to call with broadvoice from about 3 days beginning a not registered problem of, asterisk shows to a message of error with the DNS, and my dns this working fine WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) [Dec 16

Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using "BroadWorks". The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks thingie is not listening on that port, but instead on port 5060. So

Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/f6d5af5a/attachment.htm

I've been struggling with a Trixbox running Asterisk 1.6 for one of our customers as of late. The service provider in question is using BroadWorks and requires a single trunk registration for the trunk group. We have 4 users(lines/numbers) in the TG. The sip trunk is setup as follows: type=peer host=192.168.1.1 fromuser=<tgid> fromdomain=<sip domain> dtmfmode=rfc2833

Apologies if this is a repeat: I trawled through the archives and couldn't find a reasonable answer, so I'm asking here. I have an Asterisk install connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case, Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT device that connects to the Asterisk install, and using this setup I've been pretty

I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:

Hi I m trying to install a full virtualized guest on my dell poweredge 1950 with centos 5.1 64 bits. I just launched the following command virt-install -n elaphe -r 4096 -f /var/lib/xen/images/elaphe -s 40 -b vlanbr15 --hvm Would you like to enable graphics support? (yes or no) yes What is the virtual CD image, CD device or install location? /var/lib/xen/F-7-i386-DVD.iso Starting

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

OpenSSH-2.5.2.p1 won't connect to OpenSSH-2.5.1p2 using version 2 protocol, quitting with the error message: [dunlap at tesla dunlap]$ ssh -2 kraken 7a 90 3f 39 37 67 0d 9e ac 43 74 c3 83 83 f5 a2 Disconnecting: Bad packet length 2056273721. tesla is Linux tesla.apl.washington.edu 2.2.16-3 #1 Mon Jun 19 19:11:44 EDT 2000 i686 unknown Intel RHL6.2 with OpenSSH-2.5.2.p1 compiled from sources

The command "ssh ls -l /doesnotexist" gives various responses: Running from a 200 MHz PentiumPro with dsa key added to ssh-agent: Mistakes worst to fast machine: To a faster 600 MHz dual processor i686 600 MHz machine: ls: /doesnotexist: No such file or directory -- correct nothing at all -- wrong ls: select: Bad file descriptor -- wrong

I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I don?t need such data, so, anyone knows how I would configure my Asterisk box against a Broadsoft peer?

Hi Greg, it's easy: Configure searches for the original C++-Compiler, "cc". I presume you're using gcc (I do so on SUN Solaris), and it should be no problem to make configure see what it wants: Just link "gcc" to "cc" in /usr/local/bin (or whereever "gcc" is installed), and it should work, 'cause the two compilers are fully compatible. BtW: If

> Hello All, > > History ~ > I recently took a position with a call center. At the time they had > about 50 agents in a call queue. The queue was setup to ringall. The > agents use Eyebeam softphones. Everything is local lan, no routers, > everything connected via Cisco 3600 10/100 switches. > > Now we are up to about 150 agents, and I have kept everything pretty

Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071217/ae6cba5a/attachment.htm

Hello All, I have a need to connect an analog device to an asterisk server. The analog device has 4 analog lines going into it (it?s a fax solution). The fax solution answers the analog call, then listens for dtmf. The dtmf code that is played tells the fax device what email address to send the fax to. All calls on our system come into the server through a PRI. The faxes come in over a PRI,

I can see from different post that if working with sr-iov, i should work with vlan Is this an obligation to work with vlan if working with sr-iov? If not according to which parameter will the different vf get the traffic. Let's say i declare max_vfs=7, how will the traffic be seperated between the vm? However till i get an answer i tried to work with vlan And i still don't get

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp [BLUIPIN] type = aor remove_existing = yes contact = sip:bluipaddress [auth7] type = auth username =