similar to: RTP codec 13 received - Cisco incompatibility?

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

Hello, I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no success. It seems that Nuance does not send any secret/password (there is no way to define it!), this is the list of parameters that Nuance provides for registration: audio.sip.UserAgentURI=sip:user@domain audio.sip.UserAgentPort=<port> audio.sip.ProxyServerURI=sip:<IP>:<port>

Hello: I would like to use Asterisk as a redirect/proxy sip server to route SIP calls on a sip header/parameter basis. I've tried some things successfully: - SIP registration from clients. - On-the-fly compression for wan VoIP transfers: SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711 - Sending custom parameters in URI: exten => 1,1,Setvar,VXML_URL=var1=value1

Hi Cluster Community, we are seeing some problems when adding multiple terrabytes of data to a 2 node replicated GlusterFS installation. The version is 3.8.11 on CentOS 7. The machines are connected via 10Gbit LAN and are running 24/7. The OS is virtualized on VMWare. After a restart of node-1 we see that the log files are growing to multiple Gigabytes a day. Also there seem to be problems

This is strange, a couple of questions: 1. What volume type is this? What tuning have you done? gluster v info output would be helpful here. 2. How big are your bricks? 3. Can you write me a quick reproducer so I can try this in the lab? Is it just a single multi TB file you are untarring or many? If you give me the steps to repro, and I hit it, we can get a bug open. 4. Other than

A few days ago I installed dovecot stable to replace uw-imap. The install went well and all boxes were converted ok. When accessing the new imap server though, certain operations seem much slower, in particular, moving mail between boxes is very slow, and I have received several complaints from users that the mail server has slowed down. I can't figure out what the slow point is as maildir

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

Hi Cluster Community, we are seeing some problems when adding multiple terrabytes of data to a 2 node replicated GlusterFS installation. The version is 3.8.11 on CentOS 7. The machines are connected via 10Gbit LAN and are running 24/7. The OS is virtualized on VMWare. After a restart of node-1 we see that the log files are growing to multiple Gigabytes a day. Also there seem to be problems

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Hello! Is there a way to communicate from Asterisk to a TTS server? I've seen festival.conf, but it seems that it works only with Festival server. Thank you.

I'm planning on firewalling my samba server, i understand that the ports for samba are 137 139 445. Does anyone know if these are udp or tcp ports?

I have had the following problems both with the stock samba 2.2 server that comes with RedHat 9 and several releases of samba 3.0, currently it's at 3.0.14a. When logging on to Windows 2000 computers using the samba server as an NT domain controller, certain profiles seem to get permission errors and windows just sticks you with a temporary profile that changes everytime you log on. It seems

I re-added gluster-users to get some more eye on this. ----- Original Message ----- > From: "Christoph Sch?bel" <christoph.schaebel at dc-square.de> > To: "Ben Turner" <bturner at redhat.com> > Sent: Wednesday, August 30, 2017 8:18:31 AM > Subject: Re: [Gluster-users] GFID attir is missing after adding large amounts of data > > Hello Ben, >

Since my last query went ignored let me try a different approach... I've got a Telos hardware encoder that works under Shoutcast but gets "authentication errors" when trying to use Icecast. Here's what I see when trying to add the Telos as a relay: [110:Connection Handler] Kicking source 107 [192.168.200.200] [Error in request, relay refused entrance] [relay], connected for 0

Hi I want to mount the a share of a windows 2003 Domain Controler, but I'm not been able... Does any body knew a solution for that problem? Thanks.... here is the error message: <snip> root@r-pad:/mnt/max-ecom# mount -t smbfs -o username=mnicolas,password=sucker //192.168.200.99/Multimedia /mnt/ cli_negprot: SMB signing is mandatory and we have disabled it. 11688: protocol

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

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that time I heard several "pops", or "clicks". Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a "pop", instead of just silence, or a

Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware