similar to: 10.0 CallerID question

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

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

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476. Remote-Party-ID: "Cloutier"

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi, I've two yocto questions about the syntax of dialplan: 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki of Asterisk, I see very often "=>", however, what's the reason for both syntaxes authorized ? Historical ? 2. To write info in logs/console, you have two commands: NoOp and Verbose. Verbose seems to be

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/7e14a4e0/attachment.html>

Hello Everyone, Our switch is sending P-Asserted info to asterisk however the information is getting removed when asterisk forks the call. Is it possible to have asterisk retain the P-Asserted on the leg. This is quite important for valid functionality of our network. Tried setting `sendrpid = yes` and still same problem. We really don't want to have to `SipAddHeader` as it is already being

Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 Digium G.711 Licensed Channels : 1 Max Concurrent : 0 Success : 0 Switched to

I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews & Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A&A were for some odd reason not sending calls to my server, but that has been resolved. The problem now is that the calls fail to

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set "atxfer = *" (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer

It's my first post here, so I'll cut to the chase I have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the 2nd the server. The client uses sip and the server pjsip. This is the client's sip.conf [general] context = default allowguest = no