Re: [asterisk-users] what to do when a sip password includes a semicolon
asterisk-users-boun...@lists.digium.com wrote on 03/11/2016 01:43:47 PM: > From: Saint Michael> To: Asterisk Users Mailing List - Non-Commercial Discussion > , > Date: 03/11/2016 01:44 PM > Subject: [asterisk-users] what to do when a sip password includes a semicolon > Sent by: asterisk-users-boun...@lists.digium.com > > I got a new sip account, and the format > register=> user:passwrd@proxy:port > fails when the sip password has a semicolon > Is there a possible workaround? > I cannot change the password, it comes from the provider. Try escaping the semicolon with a backslash. A password of abc;123 would become abc\;123 Not entirely certain that would work, but it would be the first thing I would try. Also, I think a provider would be amenable to changing a password if it was problematic for some reason, but try the backslash first. __ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what to do when a sip password includes a semicolon
I got a new sip account, and the format register=> user:passwrd@proxy:port fails when the sip password has a semicolon Is there a possible workaround? I cannot change the password, it comes from the provider. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 devices same *actual* extension - can it be done
Kirill Marchuk wrote: And what would the behaviour be with , if a newly registered peer with same SIP login has appeared while the Dial is already progressing ? We've seen with chan_sip that there's no straightforward manner to add a newly registered peer (think mobile application that registers after receiving push notification) to a progressing (not answered) call The Asterisk dialing process itself does not allow this. Once channels are dialed you can't add. You'd need to send the push notification, wait a period of time, and then do the Dial. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users