search for: sunfon

What does the (T) mean? Am playing around with running an IAX trunk over an OpenVPN session and see this only on this peer. demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK (26 ms) Same thing on the other side: sunfone/demopbx 10.222.0.1 (S) 255.255.255.255 4569 (T) OK (31 ms) Cheers, j

...USB sticks). So hoping this will spark some comments on the concept in general, and really hoping someone has actually tackled something similar. It could open up a nice niche for even residential customers with expensive POTS lines dedicated to alarm systems. Cheers, -- Jeff LaCoursiere SunFone jeff at sunfone.com

...s soon as I loaded the kernel module in the host. Maybe this is more a question for an LXC list, but I noted that a few people had played with this (asterisk in LXC) and wondered if anyone managed to get dahdi_dummy shared across their containers... Thanks for any input, -- Jeff LaCoursiere SunFone 340-715-7600 x222 jeff at sunfone.com

...DIALOG_INFO_XML; </span><br><div><div></div><div class=3D"h5"> <br><br><div class=3D"gmail_quote">On Tue, Jun 14, 2011 at 8:36 AM, Jeff La= Coursiere <span dir=3D"ltr">&lt;<a href=3D"mailto:jeff at sunfone.com" target= =3D"_blank">jeff at sunfone.com</a>&gt;</span> wrote:<br><blockquote class=3D"= gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-= left:1ex"> <br> Struggling with an IP650 and 7 IP335s t...

...nes can handle. Look in chan_sip.c for if (strstr(p->useragent, "Polycom")) { p->subscribed = XPIDF_XML; and change that line to p->subscribed = DIALOG_INFO_XML; On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere <jeff at sunfone.com> wrote: > > Struggling with an IP650 and 7 IP335s this morning. I have the following > hints defined (courtesy of FreePBX 2.9): > > extensions_additional.conf:**exten => 300,hint,SIP/300 > extensions_additional.conf:**exten => 301,hint,SIP/301 > extensions_addit...

...ken here. And the 650 seems to "lose" its hint for a phone once in a while, and report it as unreachable, even though it can easily make and receive calls from it. Am I tilting at windmills? Is this really unstable or has someone made it work solidly? Thanks! -- Jeff LaCoursiere SunFone 340-715-7600 x222 jeff at sunfone.com

...d no effect. I watched dahdi_tool as calls came in and out but there isn't really a lot of information there. Sangoma has a cool tool "wanpipemon" that shows error stats and such on the span. Is there such a tool for Digium cards? Any suggestions? Thanks, -- Jeff LaCoursiere SunFone jeff at sunfone.com

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

...how the call ends up in this state, and what makes it finally go away. I'm assuming that in the meantime that channel on the T1 is unavailable, which is not a problem at the moment (plenty of free channels), but will surely be a problem as usage grows. Any clues? Thanks, Jeff LaCoursiere SunFone

...ou will get the "cannot be completed" message. Also ComED (electric company in the midwest) at 800-334-7661. Hit "2", then say "agent". A different voice on the "cannot be completed" message, but the effect of course is the same! Cheers, Jeff LaCoursiere SunFone

...es will ring. I'm getting confused on how the outbound will work, and am wondering if I need something in front of the two datacenters to direct outbound calls to the datacenter that is "up". But that smacks of another single point of failure. Any advice? Thanks! Jeff LaCoursiere SunFone

...ing any transmission errors, no overruns, and even the audio of 'invalid extension' seems clear before it hangs up on me. The next maintenance window I swapped out the A104D for a Digium TE122. Exactly the same issue! Some problem with Dahdi then? Any ideas? Thanks, Jeff LaCoursiere SunFone -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131209/dd9081ac/attachment.html>

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

Some !@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll number is something we can cutoff before it gets to our upstream provider? Is there some collected

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was "taken advantage of" over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN

Hi, I have several 1.4.29 installations with Sangoma AFT101d cards. Normally we have been collecting the raw data and then graphing channel use for these customers with: asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l Then I recently noticed that there were some "zombie" calls in this list that were not actually active anymore. They go

Hi, I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four port T1 card. Only one RBS T1 plugged into it right now. I have been getting complaints about random hangups. Endpoints are all remote, but I track very closely the latency (by graphing the output of "sip show peers") which normally shows me when a peer is having connectivity issues. Several that I have

I am sure this is simple, but have been struggling. I want to create a call file that dials out a particular Dahdi channel to enable call forwarding on a POTS line. I have this in extensions.conf: [custom-callfwd] exten => s,1,Answer exten => s,n,Dial(DAHDI/4-1/*717157750) exten => s,n,Verbose(${DIALSTATUS}) exten => s,n,Hangup [custom-callfwdcanc] exten => s,1,Answer exten