similar to: outbound calls not ringing still

I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.

Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to

I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via:

Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/

On 02/14/2012 11:01 AM, Jun Koi wrote: > On Tue, Feb 14, 2012 at 11:47 PM, Alex Jia <ajia at redhat.com> wrote: >> Hi Jun, >> I assume you haven't changed libvirt default URI, it may be a issue, >> I want to know whether it works for you if you explicitly specify >> --connect qemu:///system with virt-install? I think a root reason >> probably is your disk

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

Hi, I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video?mail is not stored (audio is through). Both the client?use H.264 codec with following sdp information:

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

Hi, http://llvm.org/bugs/show_bug.cgi?id=3367 http://llvm.org/bugs/show_bug.cgi?id=3831 These bugs are both cases where the optimizers are generating incorrect code. They have simple test cases and reasonably straightforward fixes. I've just done a successful test-suite run on i686-pc-linux-gnu with both fixes included. Can you please consider applying the fixes? Thanks, Jay.

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Hello On kubevirt project we are using IpV6 Kind cluster, and trying to support migration, but since both of the containers are on the same host we get this error {"component":"virt-launcher","level":"error","msg":"internal error: Attempt to migrate guest to the same host

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Jay, On Mar 30, 2009, at 10:29 AM, Jay Foad wrote: > Hi, > > http://llvm.org/bugs/show_bug.cgi?id=3367 > http://llvm.org/bugs/show_bug.cgi?id=3831 > > These bugs are both cases where the optimizers are generating > incorrect code. They have simple test cases and reasonably > straightforward fixes. I've just done a successful test-suite run on > i686-pc-linux-gnu

Hi All, We have all had to face skeptical colleagues asking if software made by volunteers could match the quality and accuracy of commercially written software. Thanks to the prompting of a recent R-help thread, I read, "R: Regulatory Compliance and Validation Issues, A Guidance Document for the Use of R in Regulated Clinical Trial Environments (http://www.r-project.org/doc/R-FDA.pdf).

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

I have the following scenario: g729 gsm UAS <-----------> * <-----------> UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ?Am I wrong? The UAC and UAS are

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CentOS Errata and Security Advisory 2018:3831 Critical Upstream details at : https://access.redhat.com/errata/RHSA-2018:3831 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: 429981c764d258970b014425dfe4a26573613bb5941d254e0deac4aa21ed0c91 firefox-60.4.0-1.el6.centos.i686.rpm x86_64: