similar to: Rejected calls to Sylantro server

Trying to make this work, and Office 365 support is useless, giving me the following response when I asked them for help troubleshooting a 488 Not Acceptable Here. Regarding your service request about configuring your PBX system with Office 365, we do not support specific setups for PBX systems for Unified Messaging. Please contact the vendor for more specific instructions and configurations.

Contacted Scott Willard at Polycom this morning, he has since been reassigned to other duties within the organization. Mr. Willard's tone seemed optimistic, and he referrred me to Roger Austin, Regional Channel Manager for Voice. Roger's reply to my inquiry is as follows: "Cory, We appreciate your interest in Polycom VoIP Phones. Polycom deploys our VoIP phones with our VoIP

I've been working with Asterisk for a little while now, and have been looking recently at my next career opportunity. It seems from searching the various job sites that the predominant VOIP technology is not the applications-based open source approach we took, but Cisco, with a really heavy emphasis on the networking (ie network engineer) aspect. If you do a job search for (VOIP or

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

So are there any plans for bridged line appearance support in Asterisk? The new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel, Sylantro etc. supposedly support it. Seems to me that Asterisk needs to get on the bandwagon or be relegated to call centers, specialized voicemail applications, and phone chat businesses. It's not needed for companies used to PBX's but

I've got a dedicated LD T1 terminating to a DMS100 switch. My outgoing calls aren't working, on the switch side they see two sets of dialing, with the first three digits repeating. I've used the sample extensions.conf modified a bit to remove the 9 and the 1, like so: [trunkld] exten => _NXXNXXXXXX,1,Monitor(wav,/tmp/dial) exten =>

I''m trying to get the hashing filters set up for about 1k IP addresses. I went through the list archives and read the howto. I did find a couple non-functional samples. Does anyone have a shell/perl script that would at least generate the basic rules for 256k, 512k, etc, and then attach the ip range to it? dave -- Dave Weis djweis@internetsolver.com http://www.internetsolver.com/

I have a machine with two routes to the internet doing load balancing on the connections and NAT for internal machines. For inbound ssh connections and outbound connections to anything, it will occasionally "lose" packets. If I Control-C out of ssh and try again, it usually works, but not always. After a few tries it does connect, but frequently stalls and hangs permanently. I did

I''m trying to shape each machine on an interface to 256k each, but I''m getting stuck and only able to shape an entire interface to 256k. What should I be doing differently here? tc qdisc del dev eth0 root tc qdisc add dev eth0 root handle 1: htb default 10 tc class add dev eth0 parent 1: classid 1:1 htb rate 100MBit ceil 100MBit tc qdisc add dev eth0 parent 1:10 handle 110:

I am located in the UK, and I am using Sipura spa-2000 adapters to connect analog phones to a voip network. The network connects to the PSTN as well via the Sipura spa-3000 adapter. I would like to provide surge protection for the spa-2000 and the spa-3000 adapters. 1. For spa-2000, fxs port: What is the maximum tip-to-ring voltage before damage to the the adapter occurs? 2. For spa-2000,

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Hello I?m need fitt growth curve with data length-age. I want to evaluate which is the function that best predicts my data, to do so I compare the Akaikes of different models. I'm now need to evaluate if changing the initial values changes the parameters and which do not allow to estimate the model. To do this I use the function nls(); and I randomize the initial values (real positive number).

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Hello together, we have very often a problem using OpenSSH v2.3.0 on Solaris 2.6. The daemon hangs up and it is not possible to login in. The -v option for ssh displays the following: # ssh -v zvadm1 SSH Version OpenSSH_2.3.0p1, protocol versions 1.5/2.0. Compiled with SSL (0x0090600f). debug: Reading configuration data /etc/ssh_config debug: Seeded RNG with 34 bytes from programs debug: Seeded

Is there anyone out there who has installed FreeBSD on the above Sony laptop ? Both ''cat filename > /dev/dsp0.0'' or ''vlc cdda:///dev/acd0@1 are OK. If I run ''cdcontrol -f /dev/acd0 play'', there is no sound. But the output of ''cdcontrol -f /dev/acd0 status audio'' is alright. (same behaviour for cd0 instead of acd0) And the output

Hello, I try to passthrough a PCI-card to a VM named testvm I want to do that with an xml-file named hga.xml including the following content: <hostdev mode='subsystem' type='pci' managed='yes'> <source> <address domain='0x0' bus='0x1' slot='0x00' function='0x0'/> </source> </hostdev> When I execute