similar to: Duplicated INVITE in SIP session?

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

I have been a reseller & subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register => 0845xxxxxxx@sipproxy.pipecall.com/1000 register => sipxxxxxxxxx:xxxxxxxxxx@sipproxy.pipecall.com/1000

Hi all! This is my VoIP network scheme H323EndPoint ----- --- GW H323/SIP-IN -- -- SIP Phone | | (Sipquest) | | | | | |

I've been looking through the examples and docs, but haven't yet quite figured out how to do what I want. We've got a T1 coming in carrying a block of telephone numbers, terminating on an Asterisk box. Any call that comes in needs to get sent to a SIP proxy, with a particular extension format: *ANI*DNIS*@sipproxy.address The closest I can see to do this with the Dial() command is:

Hello list. My phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a

Hello, I'm trying to use libvirt as a non-root user to obtain statistics on the CPU usage by VMs using the Python API. I'm performing basically the following steps: import libvirt conn = libvirt.openReadOnly(None) dom = conn.lookupByUUIDString('268e38ea-1bc7-41e4-c19e-8eff682e58e4') dom.getCPUStats(True, 0) However, they result in the following error: libvir: QEMU Driver

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

On 3/18/2018 6:13 PM, Sam McLeod wrote: Even your NFS transfers are 12.5 or so MB per second or less. 1) Did you use fdisk and LVM under that XFS filesystem? 2) Did you benchmark the XFS with something like bonnie++? (There's probably newer benchmark suites now.) 3) Did you benchmark your Network transfer speeds? Perhaps your NIC negotiated a lower speed. 3) I've done XFS tuning

Hi, I've done some similar tests and experience similar performance issues (see my 'gluster for home directories?' thread on the list). If I read your mail correctly, you are comparing an NFS mount of the brick disk against a gluster mount (using the fuse client)? Which options do you have set on the NFS export (sync or async)? >From my tests, I concluded that the issue was not

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

Howdy all, We're experiencing terrible small file performance when copying or moving files on gluster clients. In the example below, Gluster is taking 6mins~ to copy 128MB / 21,000 files sideways on a client, doing the same thing on NFS (which I know is a totally different solution etc. etc.) takes approximately 10-15 seconds(!). Any advice for tuning the volume or XFS settings would be

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi Tony, Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

Hi, As I posted in my previous emails - glusterfs can never match NFS (especially async one) performance of small files/latency. That's given by the design. Nothing you can do about it. Ondrej -----Original Message----- From: gluster-users-bounces at gluster.org [mailto:gluster-users-bounces at gluster.org] On Behalf Of Rik Theys Sent: Monday, March 19, 2018 10:38 AM To: gluster-users at

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as

Guys, please help me in understanding what I'm mistaking... Description: I've configured my AsteriskNOW (beta 6) server, in service providers section, with the parameters provided by my ITSP. Until now I've used this configuration with SJphone and all worked perfectly. Now I've decided to use this account with AsteriskNOW to begin my experience with a VoIP based PBX. The

hi i got TOS and retranssmission error on receiving SIP call chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 for seqno 43156 (Critical Response) -- See doc/sip-retransmit.txt. [May 22 13:42:44] WARNING[18021]: chan_sip.c:2821 retrans_pkt: Hanging up call 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 - no reply to

*** If anyone has a better way of doing this, please post to the list. I hadn't seen anything on this list or in channel.c/chan_local.c - which prompted this email *** I'm not sure how many VoIP providers out there are using Asterisk as a service platform like we do, but I thought I'd share an experience with call looping that was racking my brain with the list. One of the