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

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 ;

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.

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

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 have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

Hi, I've managed to get HYLAFAX---->T38MODEM----->ASTERISK---->CISCOAS5400 working, but when they are negotiating asterisk drops a message telling "Unknown RTP codec 96 received from gateway" Do somebody know how to fix it ? Thank you ! << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: Answer (4) ]

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May