similar to: Debug for SIP and reINVITES (ATA-186)

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other

is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Is there a way in iax to have to endpoints talk to each other directly (after the call is setup by *) without going through *. In sip, with * you can do it by configuring sip.conf with canreinvite = yes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030728/0c711d05/attachment.htm

I am wondering if the SIP protocol and its implementation in * allows for changing codecs mid-connection. I've seen some questions regarding this on the list, but I've not found any clear answers. I've also seen the SIP_CODEC variable, but it's not clear that it will change the codec on an existing call. Also, there are mentions of needing a reinvite to make the change, but most

Hi all, My Asterisk server is working fine, although every time that in the middle of any call there is a reinvite, the user hears a glitch. Why is this happening? How can I solve this problem? Thanks in advance, Ricardo Carvalho.

Hi All, I'm trying to understand how SIP re-invites work, if both parties are NAT'd and firewalled. I can't see how either party can initiate the conversation, and the far-end firewall would surely see any incoming packets as unsolicited, even with a STUN server giving the public IPs.. We recently moved networks, and my boss is telling me this used to work on the old network for

I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-----| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites

For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

I have been trying to do a nls model and gives me the error of a nonnumeric argument table(file="c:/tt2.txt",header=T) > fit.model <- nls(TT~60*(1+alpha*(v/c)^beta),data=tt2, start=list(alpha=1, beta=3, v=1000)) Error in v/c : non-numeric argument to binary operator > is.numeric(tt2) [1] FALSE > is.character(tt2) [1] FALSE > as.numeric(tt2) Error: (list)

Hi all: I am experiencing a problem where WinNT and Samba are dropping connections during file transfers. I have the following smb.conf settings: keep alive = 30 dead time = 1440 socket options = IPTOS_THROUGHPUT SO_KEEPALIVE (although I tried IPTOS_NODELAY with no apparent benefit). The client is copying a large number of files up to the server, then there is a pause at a random file,

I'm running into errors when using Enum lately. I can't figure out what the problem might be as I've had Enum up and running in the past. I'm running the latest CVS-Head compiled version. I've also tried using the new Enum function with the same results. When doing a lookup on a number that exists in the enum server I get the following results: -- Executing

-----Oprindelig meddelelse----- Fra: Simo Sorce [mailto:idra@samba.org] Sendt: 8. januar 2003 15:40 Til: Preben S?rensen Cc: Andrew Tridgell; Jeremy Allison; samba-bugs@samba.org Emne: Re: oplock_break (PR#26342) Please send help request to the users mailinglist samba@samba.org Simo. On Wed, 2003-01-08 at 15:38, ps@datamann.dk wrote: > Full_Name: Preben S?rensen > Samba_Version: 2.2.5

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

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 ;

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different