similar to: Asterisk/Metaswitch trunk, no inbound RTP stream on inbound calls

I'm attempting to connect to a Metaswitch, inbound only (at this time). The Metaswitch is the only "connection" (at this time). All I'm getting so far is a bunch of "OPTION" messages which my Asterisk box replies to but I don't get inbound calls. Here's my sip.conf. As you can see I've been trying a bunch of different options without success :(

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please. I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Metaswitch it goes to VM after a couple rings and the call goes to my

Hi folks, I would appreciate it if someone can help me understand some weird results I''m seeing with trying to do performance testing with an SSD offloaded ZIL. I''m attempting to improve my infrastructure''s burstable write capacity (ZFS based WebDav servers), and naturally I''m looking at implementing SSD based ZIL devices. I have a test machine with the

I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch which my Asterisk box replies to. The exchange looks like: <-- SIP read from 172.b.c.d:5060: OPTIONS sip:metaswitch@206.b.c.d:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP

Hi, all: To recap: I can successfully get my WinXP box to join the domain (WHITEROCK), but after rebooting, when I try to log on: "The system cannot log you on now because the domain WHITEROCK is not available." I get this even when using the user 'root' - and this user was the one to successfully join the domain WHITEROCK!! To rule out any stale junk, my WinXP is a

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Hello, I''m trying to port one of our existing appliances (running on 64-bit Red Hat Enterprise Linux 6.4) to a Xen HVM guest . We''re seeing some very odd behaviour that doesn''t manifest on other platforms. The guest experiences intermittent lockups of a few seconds- shell sessions become unresponsive, and various software healthchecking in our application triggers,

Im trying to get my asterisk box to register to a sip provider without much success. here is my console output in asterisk Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration for 'voicemail.nexband.com@metaswitch.nexband.com' timed out, trying again -- Got SIP response 403 "From: URI not recognized" back from 208.149.73.5 Urgent handler in my

Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

hi all, actually i have partially connected the 2 servers but there is a problem. 2 servers A and B server A forwards call to server B without any problem but when i try to forward call from server B to A, server shows the following error on the cli WARNING[7751]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is

On Wed, Apr 15, 2020 at 10:53:05 +0800, 18781374080 wrote: > > > > Hey, guys > > I've been working on whether libvirt supports encrypted snapshots,Here are my versions of libvirt and qemu > > [root@xx ~]# libvirtd -V > > libvirtd (libvirt) 4.5.0 This is too-old encrypted backing files work starting from libvirt-5.10 (but I strongly suggest using at least

Hey, guys I've been working on whether libvirt supports encrypted snapshots,Here are my versions of libvirt and qemu [root@xx ~]# libvirtd -V libvirtd (libvirt) 4.5.0 [root@xx ~]# qemu-img -V qemu-img version 2.12.0 (qemu-kvm-ev-2.12.0-33.1.el7_7.4) Copyright (c) 2003-2017 Fabrice Bellard and the QEMU Project developers 1. assign $MYSECRET to libvirt secret using the secret-define and

Hi All. I've just started using Markdown for my blog. So far it seems quite nice, but the one issue I've run into is a lack of "native" support for footnotes. I think that Daring Fireball's footnote style makes a lot of sense, and that's what I intend to use. However, it is a bit cumbersome to implement that style from within Markdown. I notice that the PHP

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk. However,