similar to: Identify endpoint based on Diversion header

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

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:

Hi all, (sending this again from the correct address) I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config. I've defined several aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................>

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

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 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] >

is there anyway to change Sip headers in local channels? if a user sets forward on their handset, calls coming in to the handset get diversion header added: Diversion: "202" <sip:202 at 192.168.1.46>;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-00000483 to 'Local/3333333333 at test' (thanks to SIP/202-00000484) and not all

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 out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Invite contains: Diversion: <sip:+41315995003 at 157.161.10.190>;reason=no- answer;privacy=off;counter=1 Asterisk Logs: RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1) From what I see in the source of chan_sip

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

I'm a bit confused about how * constructs the RURI when it generates a BYE. For the situation where * send the initial INVITE it constructs the RURI for the BYE from the contact header of the 200 OK response which is well and good. However when * receives the initial INVITE it does not use the contact header contained within to construct the BYE's RURI but constructs it from scratch. This

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

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Hi Folks, This is a totally off-topic diversion that I thought people might find fun. I've been working on a small parser framework that I'm integrating into Obversive to provide code analysis of R scripts and stuff. It is still a work in progress, but the parser currently can parse R code and produce an XML output file representing the Abstract Syntax Tree. I thought it would be

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 trying to port our configuration form sip to pjsip channel and have following issue. Sip.conf has a host parameter that sets the RURI to a given value. This functionality is needed in some of our scenarios where we need to send requests to specific IP address with specific domain in RURI. I did not found an equivalent to the host parameter in pjsip configuration. Did I

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

Hi Ian, thanks a lot for this information. I have forwarded this also to the "pkg-xen" team, because there is some real progress in bringing xen3 to debian. It might be interesting for them too. regards, Ralph Am Mittwoch, 1. M?rz 2006 16:57 schrieben Sie: > Because I didn't feel like compiling my own libc and then maintaining > the resulting system, I wrote a script to

I am just playing with a SNOM 190. Overall, I'm very impressed with the quality of the unit and the feature set. I am running the latest firmware (snom190-SIP 3.57u) and the asterisk CVS from last night (1/3/05). The only problem that I've encountered so far is with Call Forwarding, which doesn't work at all. The Snom phone is sending a "486 - Busy Here" back to *, which