search for: notransfer

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let the system a bit of time to detect if it's a fax calling, this has worked so far in all cases except today. I received a fax from overseas and it seems that Asterisk has been...

...----------- [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) tos=0x14 delayreject=yes disallow=all allow=g273.1 allow=g279 allow=ilbc allow=ulaw allow=alaw allow=gsm allow=g726 jitterbuffer=no mailboxdetail=yes notransfer=no ;register => gamcom:2046590@iaxtel.com ;register => 478933:2046590@iax2.fwdnet.net ;[fwd-gw] ;type=peer ;auth=md5 ;secret=2046590 ;username=478933 ;qualify=yes ;host=iax2.fwdnet.net ;disallow=all ;allow=ulaw #include iax_additional.conf -------------- next part -------------- [211] ca...

Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of IAX2[2109@2109]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA

...tion where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can step out of the media path and save a hop. That also works fine. However, that does not allow me to transfer someone who called my home extension at the office to someone else at the office. I have put the T/t options in t...

Hello, I have been getting outbound nufone calls dropped after about 70 seconds. CLI shows "Attempting native bridge of IAX2". I have put "notransfer=yes" in iax.conf in the main section and all identifier sections. I tried a Tt in the dialstring, but it still tries the bridge. a cvs update didn't help. internal *server1 tdm fxs pci card <--> iax2 trunk - *server2 <--> nufone - iax notrunk all g711u jitterbuffer is on...

...--Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve@daviesfam.org Sent: 22 July 2005 23:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 attempts native bridge when notransfer=yes On Wed, 20 Jul 2005, Peter Hsu wrote: > > Asterisk keeps attempting to do these native transfers.. > > Any ideas what I'm doing wrong? This is driving me crazy. Don't worry. The "attempt a native transfer" function is called, logs that it attempts it, then...

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

Hello, I've just set up DISA on my * server. I'm using it to avoid cellular overseas calling charges from support staff in the field at our customer sites. Support staff often spend hours on the phone to our UK factory. However, I'm not sure about the implications of reinvite in this arrangement. A support engineer calls in to a DID that I have from VoicePulse Connect. They match

...is routed via [A] to [B] and than back again into PSTN. Everything looks good, but, after call is answered, B performs native bridging attempt and tries to step out of voice path. And that's bad. Because of CDR's collected from [B]. On [B] and also on [A] there is "notransfer=yes" in [general] section and also in [peer/friend] definition. It probably doesn't work. I tried to use different iax2 peer for [B]->[A] call, so native bridging cannot occur. Fine, native bridging will fail, but Asterisk still writes CDR. Below is part of [B]'s con...

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

...for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from 20 tos=lowdelay notransfer=no trunk=no All calls are running as GSM, even though g.729 is also an 'allowed' codec (w/5 licenses installed). During an average call 'iax2 show channels' provides: Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 10.0.40.140 astpbx-woo...

...em , is it because of codec selection , I did try with other codecs like ulaw , the experience was same I am using asterisk 1.2.8 on RHEL4 Thanks Joseph John my sip.conf contains [666] ; Xlite Phone username=666 type=friend secret=666 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw [221] ;; Nokia E-60 username=221 type=friend secret=221 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw Send instant messages to your online friends http://uk.messenger.yahoo.com ___...

Hi to everybody, i tried to find an asnwer before posting this, but most astcc billing issues i searched refer to the case when no billing occurs at all. In my case i get only initial charges and any subsequent minute does not count for billing. In my iax.conf i entered the "notransfer = yes" but nothing changed. My test card and test calls are the following TEST-CARD en N/A N/A 6 0 0 50 ^02* TRUNK-G1 500 10 25000 Caller*ID Called Number Trunk Disposition Billable Seconds Billed Cost "3600" <3600> 02203568459 TRUNK/G1 ANSWE...

...;register to the voipbuster service register => XXXXXX:YYYYYY@sip.voipbuster.com ;Add an extension for our softphone ;Copy this and change 1234 into 1235 for a second softphone (etc) [1234] type=friend username=1234 secret=ZZZZZZ ; this is the .password. Change this !! callerid=Remko notransfer=yes insecure=very host=dynamic ;canreinvite=no context=default [1235] type=friend username=1235 secret=ZZZZZZ; this is the .password. Change this !! callerid=Remko notransfer=yes insecure=very host=dynamic ;canreinvite=no context=default ;Configure the incoming calls connection [v...

...le below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL: h...

Hi, all Here is a something I found on the web: http://www.voipbuster.com And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones instead of PC application. Did anyone try to connect astersisk and VoipBuster? Thanks, Rudolf

...o authenticate as ITSP_A iax.conf ====== [general] register => <my UserID>:<my password>@<ITSP A Server #1 domain> register => <my UserID>:<my password>@<ITSP A Server #2 domain> register => <my UserID>:<my password>@<ITSP B #1 domain> notransfer=yes bindport=4569 bindaddr=0.0.0.0 bandwidth=low disallow=all allow=ulaw allow=g729 jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes [ITSP_B] context=incoming-iax type=friend qualify=2000 host=<ITSP B #1 domain> user=<my UserID> username=<my UserID> auth=md5 secret...

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten => s,1,DIAL(IAX2/idefix) exten => s,1,NoOp(--- ${CALLERID} calling on portunity over IAX2 (${EXTEN}) ---) exten => s,2,Set(LANGUAGE()=de) exten => s,3,Ringing exten =&gt...

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like