similar to: Which features are lost when canreinvite is turned on ?

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Hi all, After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing

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

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, I've noticed that asterisk seems to stop relaying DTMFs after a call has been up for a while (~10 mins). I was just wondering whether this was intentional, or a bug. In detail here's my setup SIP Gateway --> Asterisk --> E1 --> Asterisk --> SIP Gateway The LHS gateway sends RFC2833 DTMF messages to the first Asterisk which bridges them onto the E1. They then get

Hi All, When I want use Asterisk as a PBX to cooperate SIP ITSP, I can not set the caller ID, so SIP ITSP do not accept the call. In Asterisk, I set a account in sip.conf to register on ITSP SIP Server: register => 6292@218.1.121.237/6292 And I added a user 6292 in Asterisk just like the account on ITSP SIP Server: [6291] type=friend username=6291 callerid=6291 host=dynamic

Hello list, I read on voip-info.org that Asterisk 1.4 support T38 passthrough. So I guess this means that I can have a Grandstream HT503 with T38 support and an analogue faxmachine on the other side of my Asterisk and a T38-account with a ITSP on the other side of my Asterisk machine, right ?! The fax coming from the faxmachine passes the HT503 to my Asterisk and my Asterisk sends the fax to

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Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,

I apologize if this gets posted twice. Tried once about 5 or so hours ago, and still have not seen the message on the list.... -------------------------------- I am having trouble getting DTMF mode to be set to inband on incoming calls. I have the following set, and for some reason the connection is still negotiated with rfc2833. [outbound] type=friend secret=XXXXXXX username=XXXXXXX

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine

Hi, Does anyone know the difference between RFC2543 and RFC3261? They are both SIP, but apparently incompatible. We are testing some hardware devices that support RFC3261 and it appears Asterisk is supporting RFC2543 and not completely compatible with RFC3261 (see below). Does anyone know how to configure Asterisk so it can do what is missing? 3. The problem occurs with confirm message

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

Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf

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,

Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten <-> exten calls, and not for outbound calls -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090117/a53f3178/attachment.htm

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!. I am testing firefly and I can say it's a great program, but I have a problem. When I use Sip and I activate the "canreinvite" option in Asterisk, I can't hear anything. My network is the following: -Two Firefly clients with SIP. Each firefly is in different networks behind NAT. -One Asterisk server with a public IP. First, I tested my network with canreinvite=no.