similar to: Sipura provisioning

Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones.

What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was pointed to the following: http://asteriskforum.ru/viewtopic.php?t=1761 It is in Russian, which I don't speak, but it references an Asterisk patch. Is this patch in 1.4.17? Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?) Anyone work with this?

Anyone have a chance to test any of the various iPhone SIP apps? I see there are a few out there, but most of the iTunes reviews aren't sufficiently technical to be useful. Thanks.

Hi, We have a CLI that certain users get dropped into when they log in. One of the things they can go is generate certificates (actually .p12 key/certificate bundles) that they will then scp out of the box from another host. Problem is that if their default shell isn't sh, ash, dash, bash, zsh, etc. then things break. Is there a workaround to allow scp/sftp to continue to work even for

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

On Tue, 25 Apr 2023 at 03:36, Philip Prindeville <philipp_subx at redfish-solutions.com> wrote: > > On Apr 10, 2023, at 7:24 AM, Darren Tucker <dtucker at dtucker.net> wrote: [...] > > Since you're using 9.1, the message could be an "Invalid free", since > > there was a double-free bug in that release :-( > > Forgot to ask: does this bug manifest

On 07/12/23, Philip Prindeville (philipp_subx at redfish-solutions.com) wrote: > We have a CLI that certain users get dropped into when they log in. One of the things they can go is generate certificates (actually .p12 key/certificate bundles) that they will then scp out of the box from another host. Off topic, and assuming the .p12 bundles need to be post-processed by clients for use by ssh,

On Fri, 8 Dec 2023 at 07:39, Philip Prindeville <philipp_subx at redfish-solutions.com> wrote: [...] > Problem is that if their default shell isn't sh, ash, dash, bash, zsh, etc. then things break. > Is there a workaround to allow scp/sftp to continue to work even for non-shell accounts? sftp should work regardless of the user's shell since it is invoked as a ssh subsystem

I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming

I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip

https://bugzilla.mindrot.org/show_bug.cgi?id=1856 Summary: Wrong QoS naming and obsolete defaults Product: Portable OpenSSH Version: 5.8p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: Miscellaneous AssignedTo: unassigned-bugs at mindrot.org ReportedBy:

https://bugzilla.mindrot.org/show_bug.cgi?id=1733 Summary: Enhance support for QoS (ToS) by supporting DSCP/CS and adding option Product: Portable OpenSSH Version: 5.4p1 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: ssh AssignedTo:

Hi. I have a Supermicro 5018D-FN4T (Xeon D-1541 based SBC) that I use for virtualization. I?m running Centos 7.3 on it (updated), with the CentOS-QEMU-EV.repo repository as the source for virtualization packages. I run an Ubuntu 16.04-2 guest VM on it, which is ordinary enough. What?s perhaps less ordinary is that I?ve attached a Lexar Media, Inc. ?Lexar Professional Workflow CR1 CFast 2.0 USB

https://bugzilla.mindrot.org/show_bug.cgi?id=1964 Bug #: 1964 Summary: QoS/DSCP names false translated to ToS hex value Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: amd64 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh

> On Apr 10, 2023, at 7:24 AM, Darren Tucker <dtucker at dtucker.net> wrote: > > On Mon, 10 Apr 2023 at 07:07, Peter Stuge <peter at stuge.se> wrote: >> >> Brian Candler wrote: >>>> What's odd is that the length is *always* 1231976033 (which is >>>> 0x496E7661 or "Inva" in ASCII). > > One thing that can cause this is

http://bugzilla.netfilter.org/show_bug.cgi?id=795 Summary: RELATED doesn't accommodate multicast UDP solicitation resulting in unicast reply Product: netfilter/iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component:

I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it