search for: allowtransfer

...same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes

Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well.

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97 disallow=all allow=ulaw allow=ilbc mohinterpret=default mohsuggest=default language=en useragent=TCTC PBX ;dtmfmode = info fromdomain=10.10.60.253 ;relaxdtmf=yes [15] username=15 host...

...ksys phone with forwarding enable to 2103. But is there any procedure in asterisk that we can not allow 2102 not to forward on 2103. and also i want to prevent the SIP/2.0 302 Moved Temporarily. please advice me that how can we set the user for not to forward or transfer on 2103. i have tested with allowtransfer=no in sip. Thanks in advance! Thanks, Max Alex Voip Developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/e7472c4f/attachment.htm

...11 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length: 0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600...

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

...hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on...

...callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandard: NULL ignoresdpversion: NULL allowtransfer: NULL dynamic: NULL path: NULL supportpath: NULL sippasswd: my-md5-pwd rpid: NULL domain: testers.com sippasswd2: NULL I'd greatly appreciate help on this! cheers, Olli -------------- next part -------------- An HTML...

...05,2,Set(${SIP_CODEC}=g729) // does not work exten => 2005,3,DIAL(SIP/1000 at 192.168.2.127,30,r) exten => 2005,4,Hangup ===use sipp t call asterisk 1 then forward to asterisk 2 with sip 1000. ------------------sip.conf ---------------------------------------------------- [1000] username=1000 allowtransfer=yes type=friend secret=1000 qualify=yes canreinvite=yes host=dynamic insecure=very fromuser=1000 ;dtmfmode=rfc2833 disallow=all allow=g729 ;allow=ulaw ;allow=alaw context=internal -------------------------------------------- when the calls coming from asterisk 1, i can not see any g729 inforamtion,...

...E(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer device load no result. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 10...

.... For a call from outside (e.g. via an ISDN channel) only the callee should have transfer rights. For internal calls typically both parties are allowed to transfer and for calls to the outside only the caller should have the rights. At least this is the setup I typically use. A simple flag "allowtransfer", like for the old sip channel, should resolve this. I think it is the devices that matter when creating a bridge and not the current context in the dialplan. This is an example of what is currently happening for an attended transfer when DTMF sequences are allowed Call from outside: [fr...

...on: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection: FILE * open failed! -- Unregistered SIP 'encrypted' sip.conf looks like this: [general] context=guest allowguest=no allowoverlap=no allowtransfer=no bindaddr=0.0.0.0:5060 udpbindaddr=0.0.0.0:5060 tcpenable=no tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 transport=udp preferred_codec_only=no disallow=all allow=ulaw language=en trustr...

...system back to work again. Some confs and system status: sip.conf: [1050] ; THAT'S A SOFTPHONE type=friend host=dynamic callerid=Softphone 1050 secret=xxxx context=call-center disallow=all allow=alaw allow=ulaw dtmfmode=rfc2833 canreinvite=yes nat=no qualify=yes call-limit=1 allowtransfer=yes insecure=no promiscredir=no useclientcode=no videosupport=no [7600] ; THAT'S PSTN CONNECTION username=7600 type=friend secret=xxxx qualify=no port=5060 nat=yes mailbox=7600 at default host=dynamic dtmfmode=rfc2833 context=out canreinvite=no callerid=7600 disallow=all all...

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

...------------------------ [general] language=en ; configured default language dtmfmode=rfc2833 ; default dtmfmode for sending DTMF (Dual-tone multi-frequency) directrtpsetup=no ; Disable the new experimental direct RTP setup allowtransfer=yes ; enable all transfers for peers and users match_auth_username=yes ; use 'authentication username' instead of 'username for authentication' (if available) session-timers=originate ; Request and run session-timers always session-expi...

...alls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no allowoverlap=no allowtransfer=yes udpbindaddr=0.0.0.0 tcpenable=no tlsenable=no srvlookup=yes maxexpiry=3600 minexpiry=60 defaultexpiry=120 qualifyfreq=60 disallow=all allow=alaw language=fr relaxdtmf=no dtmfmode=rfc2833 videosupport=no dynamic_exclude_static=yes canreinvite=no rtcachefriends=yes rtsavesysname=yes rtupdate=yes...

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,