similar to: Directing SIP/RTP sessions b/w UA

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi, I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine. I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is

Dear Asterisk friends, Please help me to clarify my doubt. After monitor SIP and RTP traffic with Wireshark. I found that both SIP and RTP traffic between 2 sip clients must be passed through Asterisk. Is it possible that 2 sip clients connect with each other directly for RTP session after sip session completed ? Thank you, Kamonwat -------------- next part -------------- An HTML

Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ?externaddr?, ?localnet? and ?nat=force_rport,comedia?. Asterisk registration is successful, I see in Wireshark the packets send between Asterisk and SIP server. However,

Hi! As reported earlier this week, I have problems with a sometimes-crashing Asterisk. In most of the cases safe_asterisk is able to restart it. But sometimes it crashes, so that manual interaction is necessary. The seg-faults and crashes occurs, right after call between a SIP Terminal and a legacy PSTN Terminal (PRI/Euro-ISDN), but only if the SIP Terminal hangs up as first. No problem, if the

I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before

Hi All We've been reading this in the CLI a lot lately: Received SSL traffic on RTP instance '0x7fe7481faad8' without an SSL session How can we find details about this particular RTP instance? "rtp set debug" needs an IP which is precisely what I want to know (and I don't)! Cheers Ethy

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

Hi, I'm trying to connect two asterisk instances using the method described here.. http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html under the section "Connecting two Asterisk systems together with SIP" I have an user named venu in serverA and vijay in serverB the serverA ip is 192.168.0.35 & serverB is 192.168.0.36 Here are the details of the config

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Hello list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with "register =>" statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 10000-10005 for example) - will that affect the sip sessions to sipgate.co.uk as well - or only those sessions where Asterisk acts as a sip proxy/server? Many thanks,

Hi, Anyone knows a software fax device which can act as a SIP UA? I want to have a SIP based FAX machine (sofware) on a PC associated with an Asterisk extension. Thanks, Dan

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

Dear all, I'm having great difficulty getting cross-subnet domain logon & browsing to work and have nearly reached the end of my sanity trying to figure out what's wrong. Here's my setup and what's happening (apologies if it is convoluted): Subnet A One Samba PDC with encrypted passwords. One samba file server Subnet B One Samba file server (serverB) that is the *local

I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not been able to send useful callerid info between them (callerid becomes "serverB"). serverA register statement: (serverB has the exact opposite statement) register => serverA:serverApassword at IP_of_serverB_nic/serverB users.conf of serverA: users.conf of serverB: [serverB] [serverA] type=friend

I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast. So with that assumption I imagine a platform that would not get involved with the

Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 "No proxies are up - can not send message to anyone" Xlite is not able to register to the

so, is it safe to put canreinvite=yes on a 7960? on a 1750? on a spa-x000? an xten? how the heck do i find out other than the hard way? randy -- ps: pun intended