search for: canreinvite

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?...

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 //////sip...

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...

Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send...

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. Everything was perfect, the voice quality was q...

....conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes [bv-in-1] type=friend host=147.135.8.128 context=from-broadvoice dtmfmode=inband canreinvite=no nat=yes allow=ulaw [bv-in-2] type=friend host=147.135.0.128 context=from-broadvoice dtmfmode=inband canreinvite=no nat=yes allow=ulaw [bv-in-3] type=friend host=147.135.4.128 context=fro...

Hi all, I am trying to use canreinvite in sip.conf and park/pick up calls at the same time. Problem: When I have it set up so RTP goes through asterisk (sip.conf: canreinvite=yes), # to xfer works fine. But, when I set it up so the RTP goes direct between endpoints (sip.conf: canreinvite=no), the # to xfer doesn't work. I beli...

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, correct and logical),...

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 doesn't even ring. Then I add the canreinvite=no to BOTH clients on the sip.conf, it starts to work. The problem is that all voice data goes through my asterisk server, so the delay is long...

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

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are canreinvite=yes and the PSTN termination path is canreinvite=no then calls between polycoms should not have asterisk in the media stream and wa...

...setup 7960(A)--Firewall/PAT--dsl---------Internet--------dsl--Firewall/NAT---7960(B) | | | | 7960(C)--NAT--cable----------------- -----dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C. (A) and (B) can communicate fine when both sides have canreinvite=yes. Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad th...

...m zap to sip (due to compatibility issues between my TDM400P and my Hauppauge PVR500). I've purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP phone). I managed to get it all working with my asterisk 1.4.0 installation, but I'm seeing some interesting things with the canreinvite option that I can't explain, even after reading: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite My setup: - asterisk server with: - eth0 = 192.168.254.254 (internal network) - eth1 = Internet IP-address - ZAP/1 (FXO) not used - ZAP/2 (FXS) not used - Z...

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

...allow=ulaw register => num1:pass@sip.broadvoice.com register => num2:passsip.broadvoice.com tos=0x18 srvlookup=yes nat=never insecure=yes [sip.broadvoice.com] type=peer username=NUM1 fromuser=NUM1 authuser=NUM1 secret=SECRET host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [sip.broadvoice.com.home] type=peer username=NUM2 fromuser=NUM2 authuser=NUM2 secret=SECRET host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [broadvoice-incoming] type=peer host=147.135.8.128 context=from-broad...

I've gotten canreinvite=yes to work with a sip device behind NAT!! You *MUST* port forward the SIPPort to in your gateway router to your phone. This is a MUST. Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.....

Hi all. I have an issue with T.38 and re-invites. Topology: provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension -> -> (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinvite to yes, it stop working. Do you have any idea where the issue can be? Any help will be much appreciated. Marek Soha

...0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=########## context=default dtmfmode=inband canreinvite=no disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw [kevin] type=friend regexten=1001 username=kevin fromuser=Kevin Marvin ; Specify user to put in "from" instead of callerid secret=XXXXXX host=dynamic canreinvite=no defaultip=10.1.1.16 amaflags=default ;...

...o configure external lines. extensions.conf [internal1] exten => 311000,1,Dial(SIP/teste1) [internal2] exten => 312000,1,Dial(SIP/teste2) [internal3] exten => 313000,1,Dial(SIP/teste3) [teste1] sip.conf [teste1] type=friend username=teste1 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal [teste2] type=friend username=teste2 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal2 [teste3] type=friend username=teste3 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal3 -- Ralph Liebessohn ICQ: 74835911 Skype: liebe...

...MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = "test" <8000100> callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line => aaln/1 callerid= "test" <8000101> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/2 callerid= "test" <8000102> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmf...