Re: [asterisk-users] Rebooting a Grandstream

Well, it doesn't seem to work on my GXP2000's!Is there a configuration options that I need to set? TIA, Mike. On Thursday 21 July 2011 6:59:19 pm Alec Davis wrote: > That works for us with GXP2000's and GXP2010, but not the later HD series > GXP21XX. > > Alec > > > -Original Message--

Re: [asterisk-users] Rebooting a Grandstream

That works for us with GXP2000's and GXP2010, but not the later HD series GXP21XX. Alec > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Mike Diehl > Sent: Friday, 22 July 2011 10:50 a.m. > To: asteris

[asterisk-users] Strange network issue

Hi all, I've got a strange problem with a customer's phones. They've got a bunch of Grandstreams that seem to be rock solid... until 7:00pm. At 7:00, some of the phones become unavailable, and stay down. Call quality is solid almost all the time. But right at 7:00, things go bad. Only some

[asterisk-users] Per-line registration

Hi all, I'm trying to figure out how it is that a couple lines on a given phone, with 3 lines, can qualify as unavailable while the remaining lines can be available. I've got qualify=1000 in my sip.cfg. Shouldn't this be an all-or-nothing proposition? -- Take care and have fun, Mike Diehl.

[asterisk-users] Rebooting a Grandstream

Hi all, I've got a number of Grandstream phones and I'd like to be able to reboot them remotely, as I do my Polycoms... I've got this in my sip_notify.cfg: [grandstream-check-cfg] Event=>sys-control Doesn't seem to work. Any ideas? -- Take care and have fun, Mike Diehl. -- ___

Re: [asterisk-users] FAX with SIP

On 11-07-21 05:52 PM, Joaquin Sosa wrote: On Thu, Jul 21, 2011 at 17:39, Kevin P. Fleming wrote: We do this in our testing all the time, and it works fine. Since you didn't specify any particular version of Asterisk, there's no way to associate your "It won't work" statement with anything in pa

Re: [asterisk-users] FAX with SIP

On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming wrote: > On 07/21/2011 04:43 PM, Israel Gottlieb wrote: > >> >> >> On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming > > wrote: >> >>On 07/21/2011 04:34 PM, Joaquin Sosa wrote: >> >>On Mon, Jul 18, 2011 at

Re: [asterisk-users] FAX with SIP

On Thu, Jul 21, 2011 at 17:39, Kevin P. Fleming wrote: > We do this in our testing all the time, and it works fine. Since you didn't > specify any particular version of Asterisk, there's no way to associate your > "It won't work" statement with anything in particular. Given the variations > of T.3

Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

On Sun, Jul 10, 2011 at 04:11, Tzafrir Cohen wrote: > > > The solution: IOMMU: http://en.wikipedia.org/wiki/Iommu . > The CPU of the system has a Memory Management Unit (MMU) that > maps virtual address spaces to processes. Likewise we know prevent the > IO card from seeing physical addresses. Rat

Re: [asterisk-users] FAX with SIP

On 07/21/2011 04:43 PM, Israel Gottlieb wrote: On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming mailto:kpflem...@digium.com>> wrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesmailto:davies...@gmail.com>> wrote: The magic

Re: [asterisk-users] FAX with SIP

On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming wrote: > On 07/21/2011 04:34 PM, Joaquin Sosa wrote: > >> On Mon, Jul 18, 2011 at 07:58, Steve Davies wrote: >> >>> The magic sauce that you need is "T.38" - Asterisk 1.6 supports this >>> to a limited degree, and your ITSP will need to support i

Re: [asterisk-users] FAX with SIP

On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Davies wrote: The magic sauce that you need is "T.38" - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the informatio

Re: [asterisk-users] FAX with SIP

On Mon, Jul 18, 2011 at 07:58, Steve Davies wrote: > The magic sauce that you need is "T.38" - Asterisk 1.6 supports this > to a limited degree, and your ITSP will need to support it. > > The sip.conf.sample file and the voip-info wiki has all the > information you need to try it out. > Correct.

Re: [asterisk-users] My Asterisk Box was hacked

Are you sure your box was actually hacked? Or did someone take advantage of a configuration error? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] asterisk's SDP

On 07/21/2011 03:54 PM, vip killa wrote: What if asterisk sends telephony events that are not in range of 0-15 though? You are misunderstanding how SDP works; when an SDP offer or answer is sent, that indicates what the sender is willing to *receive*, not what it is going to send. If the So

Re: [asterisk-users] asterisk's SDP

What if asterisk sends telephony events that are not in range of 0-15 though? On Thu, Jul 21, 2011 at 4:47 PM, Kevin P. Fleming wrote: > On 07/21/2011 03:30 PM, vip killa wrote: > >> We have a peer (a Sonus Media Gateway), that sends "a=fmtp:101 0-15" >> Asterisk sends "0-16" back, is there anywa

Re: [asterisk-users] asterisk's SDP

On 07/21/2011 03:30 PM, vip killa wrote: We have a peer (a Sonus Media Gateway), that sends "a=fmtp:101 0-15" Asterisk sends "0-16" back, is there anyway to have asterisk send a 0-15? No, and it's completely unnecessary. Asterisk is willing to accept telephony-event codes 0 through 16, but the

[asterisk-users] asterisk's SDP

We have a peer (a Sonus Media Gateway), that sends "a=fmtp:101 0-15" Asterisk sends "0-16" back, is there anyway to have asterisk send a 0-15? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asteri

Re: [asterisk-users] My Asterisk Box was hacked

On Thu, 21 Jul 2011, Robert Huddleston wrote: When I get hacked I typically run a rootkit checker http://www.chkrootkit.org/ How often do you get hacked? How are 'they' breaking in? -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] My Asterisk Box was hacked

When I get hacked I typically run a rootkit checker http://www.chkrootkit.org/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace Sent: Thursday, July 21, 2011 2:18 PM To: asterisk-users@lists.digium.c

Re: [asterisk-users] My Asterisk Box was hacked

On Thu, 21 Jul 2011 13:29:09 +0800 Malvin Rito wrote: > My asterisk box was hacked! Can anyone help on how do I secure my > asterisk box, currently my box is installed with 2 NIC. 1st NIC is > for LAN access and 2nd NIC has a public IP which is registered to our > VoIP Provider. Seven Steps to

Re: [asterisk-users] Functions not autoloading

Just received a call and on checking messages I now see: ERROR[14824] pbx.c: Function MASTER_CHANNEL not registered Grrr, looks like time to go back to 1.8.3 as all the apps and functions exist in /usr/lib/asterisk/modules. How could I help to debug this please ? -- Thanks, Phil - Origina

Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, July 21, 2011 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functi

Re: [asterisk-users] Functions not autoloading

On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote: Since upgrading to 1.8.5.0 I have had to add into modules.conf: load => func_callerid.so load => func_cdr.so otherwise they do not get loaded even though I have set autoload=yes. Is this something you would expect as it is different behavior to 1.

Re: [asterisk-users] My Asterisk Box was hacked

Really, since you sound like a novice in the Asterisk world, maybe rolling your own solution isn't a good idea. Why not use an all-in-one solution like PBX in a Flash? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Be

[asterisk-users] Functions not autoloading

Since upgrading to 1.8.5.0 I have had to add into modules.conf: load => func_callerid.so load => func_cdr.so otherwise they do not get loaded even though I have set autoload=yes. Is this something you would expect as it is different behavior to 1.8.3.0 and I do not see any issues in /var/log/as

[asterisk-users] Asterisk doesn't like OpenBTS!!!

HI list, I have succeeded to establish calls using OpenBTS/USRP1/Asterisk. but the problem is that my cell phone rings, I get 2 way audio but after a few seconds the call is dropped. In my asterisk log I see this: [Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3622 retrans_pkt: Retransmission timeout

Re: [asterisk-users] AsteriskNow install addons despite license conflict with FFA and G.729

Any suggestions on how to install Asterisk addons despite the license conflict? BTW, on another system, we installed the addons first and then the paid licenses from digium and there was absolutely no problem running and installing both. It seems to happen only when Digium software is installed BEF

Re: [asterisk-users] a=sendonly Music On Hold ignored

On Tue, Jul 19, 2011 at 10:12 PM, Matthew J. Roth wrote: > Michael wrote: > > > > True. In the working system, LAN calls are also using G.729, while > > in the non-working system, LAN calls are in G.711 (supported but > > not prioritized by the phones) and only the SIP trunk to the ITSP > > is se

[asterisk-users] Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)

Hi all We use a Genband Safari C3 Softswitch and have an attached Asterisk for some special funktions like SPOT Filtering. Now then a call comes from PSTN to a SIP subscriber, the invite looks like: From: ;tag=7f33ff47+1+68530003+e1367280 If the device is a Snom M9 (or many others) in the abse

Re: [asterisk-users] Multiple SIP trunks between same pair of asterisk box

If it is just matter of billing you can pass billing related info in additional SIP headers on single trunk. If you must need multiple trunk you can add multiple IPs of different subnet class to both interfaces and configure asterisk to listen of all IPs. Then use one trunk per IP Subnet class.