similar to: Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

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

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 List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same

Hi All, When receiving an invite containing two different caller ID, one in FROM header and the other in "P-Asserted Identity" Header, Which one will be used by the callee ? I couldn't find any RFC specifying this detail. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at

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

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 Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

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

Hi List One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls routed to Customers need to be put in the 'local' format. 0615995555 This is also the format of the From / To / Invite header

Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Hello, Is there any way to identify an incoming session based on the Diversion header? In my scenario, I have some unregistered endpoints (mobile phones) that make calls through our Asterisk, which controls the external call rights based on the endpoint's context. In a normal call, their number will be in the From header and the destination in the To an RURI, but when they make a call

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Hi Richard Thank you > You need to set more redirecting information [1]. > > In sip.conf send_diversion=yes needs to be in effect. You also need > to setup > the from party id information (at least the from number) to indicate > where you > are redirecting from. You should also increment the redirecting > count. > > Richard > > [1] >

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

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