similar to: Asterisk 1.8 Transfer CallerID

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Another dumb conversion question (I hope). I installed 10.0 and copied my 1.4 configuration files over. With a few tweaks everything works great except for 1 feature that I specifically went to 10.0 for. When I do an attended transfer, I still get the receptionists caller ID on the transferred phone instead of the incoming callerID. My assumption is that there is some

Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hello list, whenever I press the #-key I hear a voice saying 'transfer'. How can I use the #-key without this voice-message or without having it the function of unattended transfer ?! The T or t option is not set in my Dial()-command so I don't know where this transfer is coming from in the first place. Kind regards, Jonas. -------------- next part -------------- An HTML

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. Here is a link to pastebin with the Sip trace. In it you

Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: > Hi, > >

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Hello list, Asterisk 1.6.16.1 I have the following registrations : register => 119909:passwd at sip.prov.org/52525252 register => 119909:passwd at sip.prov.org/59595959 [119909] type=friend host=sip.prov.org username=119909 defaultuser=119909 secret=passwd context=TRUNKin extensions.conf : [TRUNKin] exten => _52525252,1,NoOp(context TRUNKin - 52525252) exten =>

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Release 0.66 of IPSwitchBoard is now available for FREE download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Enhancements: Support for Call Parking and retrieve/forward them again. Last Call on the Queues Page now displays a date-time in human readable format. Added CallerID on the Queue Members listing on the Queue page. New page with Agent information. Minor bug

Hello. I notice that when a call that is recorded with MixMonitor is transfered to another co-worker, the recording ends. exten => 409,n,Macro(SDstartrecording,external,${DID}) the incoming call then goes to a queue... [macro-startrecording] ; ARG1 = incoming DID or CALLERID(name) ; ARG2 = outgoing dialnumber ... exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this