similar to: SIP app for iPhone that works well with Asterisk?

That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En

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

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'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

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 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

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

> 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

Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good

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:

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

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

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:

For the 3rd consecutive term, the US Senate has introduced the "Truth in caller ID Act of 2009". It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out

I would like to setup an iphone to be an extension on my pbx. I have looked at SIAX as well as Asteriskc2d. Does anyone have any experience with either of these or another app? The important thing for me is that I can run it in the background so I can always be available to receive a call. It has to work over both 3g as well as wifi so I can take advantage of my data plan and not use minutes

Hi,