similar to: PJSIP and P-Asserted-Identity

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var:

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

Hi, I am looking for some help on dict configration. I'm trying to config shared and public mailbox, and use sql as acl backend. Here is my config: table: CREATE TABLE `shared_dict` ( `id` int(10) NOT NULL AUTO_INCREMENT, `from_user` char(50) NOT NULL, `to_user` char(50) NOT NULL, `status` int(10) NOT NULL DEFAULT '1', PRIMARY KEY (`id`), KEY `from_user` (`from_user`),

Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03-03 00:18:58]

Hey there folks, I''m sure there is a stupidly easy answer for this, but I can''t seem to find it. I''m building a site and I want a way for one user to contact another. Simple enough. The problem lies in the fact I want the email to ''come from'' the user, and not from my system. In my ActionMailer class if I do the following: @from =

Hi, In PJSIP configuration, I thought that "from_domain" parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain. I've tested at the transport level, I see no changes. I've also tested with realm parameter in auth configuration, it seems to change only the digest auth value during registration. I'm pretty new with PJSIP,

Hello, Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I got the "from" to be the way I need it. From: <sip:e04f43a2ed59 at xaccel.net;tag=44l1nRmW2 To: "TEST" <sip:5tf2f2s0rbtdj-20d14fl6n65t0o-0u03 at 34.221.174.202> I have tried a lot of changes to get to this but nothing works. I am getting this From: sip:109643183 at

Hi! We run dovecot 2.1.6 with ACL support and MySQL dict for shared/shared-boxes/user/$to/$from. When a user does a SETACL (via IMAP) to himself he is added as $to and $from to the MySQL dict. Se for example this IMAP conversation: 5 GETACL INBOX * ACL "INBOX" "user at domain" lrwstipekxacd 5 OK Getacl completed. 6 SETACL INBOX user at domain lrwstipekxacd 6 OK Setacl

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip

On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best

On Fri, Nov 02, 2018 at 09:14:51AM -0700, Linus Torvalds wrote: > On Fri, Nov 2, 2018 at 6:04 AM Michael S. Tsirkin <mst at redhat.com> wrote: > > > > I've tried making access_ok mask the parameter it gets. > > PLEASE don't do this. Okay. > Just use "copy_to/from_user()". Just for completeness I'd like to point out for vhost the copies are

On Fri, Nov 02, 2018 at 09:14:51AM -0700, Linus Torvalds wrote: > On Fri, Nov 2, 2018 at 6:04 AM Michael S. Tsirkin <mst at redhat.com> wrote: > > > > I've tried making access_ok mask the parameter it gets. > > PLEASE don't do this. Okay. > Just use "copy_to/from_user()". Just for completeness I'd like to point out for vhost the copies are

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote: > > Thanks George. > > I will correct my local_net in the morning. > > Vitelity chan_sip settings I have working, do not have a fromuser. > sip.conf settings... > > I think you can actually specify anything, it just has to be populated with something other than a sub-account username. >

Hi all, I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so I can use the GW. For this I use: exten => s,1,NoOp(One of our workers (${CALLERID(number)}) is calling office) ;callerID is the one of the calling mobile phone exten => s,n,Background(silence/1) ; Nokia E65 send digits in

Hey guys and gals, I have a snippet of code: Net::SMTP(@host, @port, @from_domain) do |smtp| @emails.each do |email| begin smtp.send_message email.encoded, email.from, email.destinations @emails_sent += 1 rescue Exception => e # blah end end end What I want to do is: Say there are 4 emails. First email is sent OK On the second email smtp raises a IOError

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >>

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