similar to: RTP codec 13 received - Cisco incompatibilit y?

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

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

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

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.

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

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

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi All - First post, and something of an audio newbie, so please be gentle. I'm cross-posting this to both icecast and vorbis, albeit in separate emails. I'll try to give as much detail as possible so this might get a bit wordy. I've been trying to get a live stream running for a local college radio station. It's currently up on a Slackware system - but

Hi All - First post, and something of an audio newbie, so please be gentle. I'm cross-posting this to both icecast and vorbis, albeit in separate emails. I'll try to give as much detail as possible so this might get a bit wordy. I've been trying to get a live stream running for a local college radio station. It's currently up on a Slackware system - but

Hi, I updated my two samba DC's from 4.0.3 to serner 4.0.7. Both servers run debian wheezy and the add was created at the beginning of the year with an classic upgrade to version 4.0.0. Recent release notes do not provide information about required upgrade tasks. So i ran. samba-tool dbcheck --reset-well-known-acls. On the first DC it found a few errors about missong members in computer

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

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