similar to: canreinvite per route

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

Hi, Does anyone had the problem of asterisk SendFax + spandsp sending only the first page of a multi-page TIFF file? Seams to be related to spandsp ECM config. Any thoughts about it? Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121005/ac471600/attachment.htm>

Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho,

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 & 2002 are behind one firewall, and 2003 & 2004 are behind another. Tim

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

Hi all, Does anyone uses astDB for a large amount of data, in special for implementing black lists with millions of numbers (i'd like about 2 or 3 million)? That would be held in memory right? Is this (memory consumption) the only problem I could face? Att. Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hi, I've noticed an unusual behavior on the dialplan execution: assume this DP: exten => _6XXX,1,NoOp(test1) exten => _XXXX,1,NoOp(test2) exten => _XXXX,2,NoOp(test3) If I call 6000 then test1 and test3 NoOps get executed, even though the pattern is different. I've always thought that if I call 6000 it would match the 6XXX pattern, that only has 1 priority, that would get

Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you -------------- next part -------------- An HTML

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Hi all, After * starts the command "queue show" would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any "QueueMemberStatus" for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a

Hi all, When I have 2 masks that would like to execute the same logic, there is the way to use the Goto (or any other) command without changing the ${EXTEN}? Eg. DID range is 1200-1349 -> call Macro(disca), what mask to use? (I just got it with 2 masks, but I didn't wanted to duplicate the dialplan for both) [test] exten => _12XX,1,Set(DIR=3) exten =>

Hi all, I'm trying to use the CDR() function on the "Variable" header of the Originate AMI action, but it isn't working. Anyone knows anything about this problem? asterisk 1.4.26 Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed

Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David