Re: [asterisk-users] ST2030 replacement

2016-01-07 Thread Frank
On Thu, 2016-01-07 at 17:35 +0100, Sil wrote: > Can you give me a return on the models you use ? Yealink T26P -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] placing calls with linphone.org SIP account

2016-01-07 Thread Frank
On Wed, 2016-01-06 at 11:27 +0100, Yves wrote: > how can I call other users > registered at other SIP-Providers? > I tried all well-known SIP URI Syntaxes but none worked... does anyone > reliably know, if it is possible at all and if so, what is the > dialstring looking like? It depends if

[asterisk-users] ST2030 replacement

2016-01-07 Thread Sil
Hello, I am looking for a replacement for my Thomson ST2030SIP. My specifications are as follows : - 2 lines. - 6 BLF keys. - PoE. Can you give me a return on the models you use ? Thanks. Sil -- _ -- Bandwidth and Colocation

Re: [asterisk-users] ST2030 replacement

2016-01-07 Thread Markos Vakondios
Grandstream GXP-1628 On 8 January 2016 at 01:00, Frank wrote: > On Thu, 2016-01-07 at 17:35 +0100, Sil wrote: > > > Can you give me a return on the models you use ? > > Yealink T26P > > > > > -- > _ >

Re: [asterisk-users] The To header was truncated in call... Whats this means?

2016-01-07 Thread Yves
I have seen these messages only on asterisk boxes that are open to public and I think this may have something to do with sip-attacks... I´d recommend some wiresharking or at least sip debugging... yves Am 07.01.2016 um 21:23 schrieb Vitor Mazuco: Hi everybody, My Asterisk, all time appear

[asterisk-users] The To header was truncated in call... Whats this means?

2016-01-07 Thread Vitor Mazuco
Hi everybody, My Asterisk, all time appear this log [Jan 7 15:37:04] ERROR[1174] chan_sip.c: The To header was truncated in call '6c66e5b6058ae257003c0f7e778da0fe@191.x'. This call setup will fail. [Jan 7 15:37:18] ERROR[1174] chan_sip.c: The To header was truncated in call

Re: [asterisk-users] Getting Asterisk to use the SIP Path header

2016-01-07 Thread Peter Baines
If anyone else is having this issue, asterisk 1.8.32.3 uses the Path header as expected, if you want to follow progress I've created a bug report: https://issues.asterisk.org/jira/browse/ASTERISK-25666 On 6 January 2016 at 10:29, Peter Baines wrote: > Hi, > > How do I get

Re: [asterisk-users] Virtual domain redirects

2016-01-07 Thread D'Arcy J.M. Cain
On Thu, 7 Jan 2016 00:04:05 -0500 "D'Arcy J.M. Cain" wrote: > I am trying to figure out how to allow da...@example.com to be > translated to dc2...@vex.net (out ISP domain) but I am at a loss to I think I see where I can hook this. same => n,Verbose(0,To: ${SIP_HEADER(To)}) This