[asterisk-users] Asterisk registers with TLS, but sends out calls via UDP

Hi, I have an Asterisk 13.8.2, which is supposed to be only a client to an encrypted SIP service. All local phones are connected via UDP. Since I can't use PJSIP (see my mailing list post from yesterday), I tried configuring chan_sip to work that way. My settings are: [general] context=public

Re: [asterisk-users] T.38 with Audiocodes gateway [SOLVED]

2016-05-03 16:43 GMT+02:00 Matt Fredrickson : > On Fri, Apr 29, 2016 at 1:34 AM, Olivier wrote: > > Hello, > > > > I'm helping a colleague (*) which has the following setup: > > > > ITSP --- --- Asterisk 13 --- -- > > Audiocodes MP-112 --- --- Fax

Re: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6

- On May 4, 2016, at 8:49 AM, Mamadou NGOM n...@numericap.com wrote: > Hello everybody, > When I call my extension the agi script don't work well. when I look at the > cli, > that is what I have: > AGI Tx >> agi_request: **.php > AGI Tx >> agi_channel: SIP/myprovider-0007 > AGI Tx

Re: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6

On Wednesday 04 May 2016, Mamadou NGOM wrote: > Hello everybody, > When I call my extension the agi script don't work well. when I look at > the cli, that is what I have: > [stuff deleted] > AGI Tx >> agi_arg_1: 56 > AGI Tx >> > AGI Rx << SET VARIABLE ** 2 > AGI Tx >> 510 Invalid or

[asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6

Hello everybody, When I call my extension  the  agi script don't work well. when I look at the cli, that is what I have:AGI Tx >> agi_request: **.phpAGI Tx >> agi_channel: SIP/myprovider-0007AGI Tx >> agi_language: frAGI Tx >> agi_type: SIPAGI Tx >> agi_uniqueid: ***AGI Tx >>

[asterisk-users] Anyone have problems with HPE 5130 EI Switch Series

Have a strange issue at a customer, they went and replaced all of their old PoE switches with brand new HPE 5130 EI Switch Series. Their PBX has been up and stable for several years with no recent changes, but since they change the switches they are having a problem with some their Yealink t-26

[asterisk-users] Double queue calls being delivered to agents

Sorry for last post -- forgot to wipe out the digest contents :/ Derek B -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Double queue calls being delivered to agents

Awesome. Thanks again Richard. On May 4, 2016, at 10:59 PM, Richard Mudgett > wrote: On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett > wrote: On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski

Re: [asterisk-users] Double queue calls being delivered to agents

On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett wrote: > > > On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski > wrote: > >> I posted this over in asterisk-dev, realized I probably should have put >> it here. >> >> Hi there, >> We’ve been having a

[asterisk-users] Double queue calls being delivered to agents

was missing. The > new default "no" doesn't send 180 Ringing any more ... . > >> Since "never" was >> the default, but most users probably expect "no" this patch updates the >> default for the "progressinband" setting to "no." >

[asterisk-users] Asterisk 1.8 secure SIP session only

Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I keep getter an error, == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open

Re: [asterisk-users] Asterisk 1.8 secure SIP session only

Your CA cert is missing. Add in sip.conf: tlscafile=/etc/asterisk/keys/ca.crt You don't need: tlscapath=/etc/asterisk/keys On 4 May 2016 at 19:43, Motty Cruz wrote: > Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. > I keep getter an error, > >

[asterisk-users] UAC and UAS for timer refresher header

Hi all, I have a intriguing issue that the RFC is not really clear about. Sometimes call hang-up on 45min mark because no-one refresh the call ( far-end hangup) On both good and bad calls: 1)We initiate an invite 2)200OK is answered as refresher=UAS 3)Send ACK 4) 'follow-up' re-invite is

Re: [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more

On 05/03/2016 at 09:16 PM Joshua Colp wrote: > Eric Wieling wrote: >> I don't know the default setting for progressinband in the code, but it >> is documented in Asterisk 11's sip.conf.sample as defaulting to never. >> Maybe the docs were fixed since Asterisk 11. > > The behavior change to