similar to: OT - P-asserted-identity and remote id

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

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

I am working with a customer and their SIP provider is IPitimi. The customer needs to sometimes provide various caller id number for the calls going to IPitimi. They are processing calls for multiple businesses who want their caller id to show up. When no caller id is provided, the From must be the DID at ipitimi ip address and caller id is DID at customer IP address. When caller id is

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

For INBOUND calls, does Asterisk support P-Asserted-Identify or Remote-Party-ID, or does it support both? Again, this is for INBOUND only. I know how to add those headers for outbound calls. My guess from what I have seen is that it supports both, but I wanted to check with the list. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

Hi All, We have two servers, one server (SIP asterisk server) sending calls to the second server(has PRI) which goes our through the PRI's (using TE 412p). When the pprivacy is enabled: P-Asserted-Identity Header, privacy "id" are sent in the header of SIP invite packet to the second server, how can we identify this privacy and block the callerid as the call goes to the second

Hi All, I got this figured out, when the privacy is ON at the other end of the server and when we get the Invite message to the server connected to PRI's, just take the details from the invite message in the Dial plan and send the calls as anonymous: exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)}) exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =

We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number. Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating? Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at

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

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

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>

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

Hi, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g.