similar to: SIP registration without password (secret)

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, 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 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

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.

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

Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -,

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 We are currently using Asterisk 1.2.4 with IAX and app_meetme for conferencing, but are looking to move to SIP because of issues with an IAX control we're using. The reason we moved from SIP to IAX in the first place was because of the poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older

----- 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 >

Hi, I installed samba4 (zentyal 3.4) and I followed the Samba4 AD DC Howto from the wiki to test the linux-side. There all seems OK. Unfortunately, when I try to register a Win7 PC, the domain is not found. So what can I do to test things on the Windows-side ? I did : C:\net view /domain:ace_domain returns : \\ZENTYAL1 which is my DC After manually "mounting" the test1-share : C:\net

Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: "http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html", but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve

Hi. there is a way to MARK udp VOIP (SIP) traffic, in order to put in a highest prio class ? Traffic flow seems start on udp 5060 port, but next both server and client seems jump to a random(?) port. I can''t use CONNMARK because is udp traffic. I only see a pattern for L7 patch in order to SIP traffic identification , but I run 2.4 kernel series . When you patch 2.4 kernel with

I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake -------------- next part -------------- An HTML attachment was

Hi, I installed a program and an error occurred: *** ERROR: An error occurred while registering the file 'C:\windows\system32\PrintPreview.ocx' *** ERROR: (User Responded with 'Abort') *** DURING THIS ACTION: DllSelfRegister: "C:\windows\system32\PrintPreview.ocx" Is there any way to register this file? Thanks. -- Sávio Martins Ramos - Arquiteto Rio de Janeiro

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",