similar to: "No Authority Found"

Hi, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange Anyhow I was getting an error: Process_sdp: No compatible codecs! And from the SIP

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

Does anyone know if this unit works with Asterisk? Thx. B.

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

Does anyone have a copy of spc.exe they could send me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error

Hello all, I am having trouble getting my IAX based Voip provider setup. Any pointers are welcome. So here is the deal. I am registered up and I can make outgoing calls but incoming calls fail. Configs all look good I thought. My PBX is behind our firewall with a direct NAT of one to one for an external IP. IAX port is forwarded UDP and TCP to the internal IP. * shows good registration and

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN => Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I

Hi, Has anyone else experienced problems as of the last couple of months when outbound calling through Freshtel? I've started getting a "No authority found" error. I've tried contacting them, and they seem to have some serious communication issues with their IT team, infact I think they have serious issues in their IT team full stop. First they can't find my account in

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true

https://bugzilla.netfilter.org/show_bug.cgi?id=1673 Bug ID: 1673 Summary: bug egress hook virtio interface with VLAN Product: nftables Version: 1.0.x Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: kernel Assignee: pablo at netfilter.org

https://bugzilla.netfilter.org/show_bug.cgi?id=1672 Bug ID: 1672 Summary: bug egress hook virtio interface with VLAN Product: nftables Version: 1.0.x Hardware: All OS: other Status: NEW Severity: normal Priority: P5 Component: kernel Assignee: pablo at netfilter.org

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm