[asterisk-users] Problem installing B410P BRI card for asterisk

dahdi_tool[40+3000] And a lot of "Wrote 0x0 to register 0x1ab but got back 0x4" statements. If i run dahdi_tools it fails with a segmentation fault. Any suggestions are appreciated! Kind regards, Marco Mooijekind. -- ___

[asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds

, version of LibPRI etc. has anybody experienced these problems on BRI? Any suggestions with regards to these warnings are welcome! Kind regards, Marco Mooijekind. -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] SuSE Firewall2 - Port Forward Command

SEfirewall2 For example: FW_FORWARD_MASQ="0/0,192.168.10.1,udp,5060,80,192.168.2.3" lets you able to forward the udp 5060 from the IP 192.168.10.1 to 192.168.2.3 You need to add all the other RTP relevant rules. Best regards. Marco Signorini -- = EtherMa

Re: [asterisk-users] Problems with Dahdi 2.3.0.1 trying to load OSLEC

Best regards, Marco Signorini. -- = - http://www.ethermania.com - - http://www.ingegnitech.com - Jose P. Espinal wrote: > Hello list, > > > I'm facing a little issue with dahdi attempting to load the OSLEC echo > canceller into my current ke

[asterisk-users] SIP CANCEL, Reason

Hi All! i have a problem with asterisk 1.8.11. I must have in the SIP cancel message, the line "Reason" Example : Reason : SIP;cause=16;text="Normal Call Clearing" I have already enable "use_q850_reason=yes", but this not work. In my dialplan I have already add : exten => _X.,n,Hangup(${HANGUPCAU

[asterisk-users] R: SIP CANCEL, Reason

[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan Inviato: giovedì 20 settembre 2012 13:42 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] SIP CANCEL, Reason - Original Message - > From: "Marco Colombo" &

[asterisk-users] Asterisk and History-Info

Hi All, Someone can tell me if asterisk support the SIP History-Info? If it supports, how can enable it? I searched on Google, but I could not find anything... Thanks for all Best Regards MC

[asterisk-users] R: Asterisk and History-Info

oun...@lists.digium.com> [mailto:asterisk-users-boun...@lists.digium.com]<mailto:[mailto:asterisk-users-boun...@lists.digium.com]> On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:33 AM To: Asterisk-Users Subject: [asterisk-users] Asterisk and History-Info Hi All,

[asterisk-users] R: R: Asterisk and History-Info

IONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "+39zzz" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 239 Thanks a lot! Marco Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-

[asterisk-users] R: R: R: Asterisk and History-Info

Ok, thanks for all Best Regards Marco -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Joshua Colp Inviato: mercoledì 26 settembre 2012 19:37 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto

[asterisk-users] Call Hold problem

lds is present the local ip address and not the next hop ip. This is the log : http://pastebin.com/ARUC0j4t The asterisk IP : 87.248.56.101 The next hop IP : 87.248.56.100 Is it a bug? i'm already search on google, but i dont find anything. Let me

Re: [asterisk-users] DAHDI and Oslec

#x27;s very old so I can't tell you if this is something true for Debian 6.06 too. Thanks. Marco Signorini. On 02/26/2013 05:38 PM, Doug Lytle wrote: I'm hoping someone can help me here. I've purchased replacement systems for 3 aging 1.4.x installs. I'm hoping to setup

[asterisk-users] Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6

Hello I recently heard this should be possible. Has anyone experience with this? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every T

Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

Hi Did you looked at Arduino + Ethernet Shield? Is something you can program in C or C++ to receive a simple TCP and/or HTTP packet and turn on an external relay. >From the dialplan you can run an http query through curl and/or an external AGI command. Best regards, Marco Signorini. -- Ma

[asterisk-users] sipML5, Ast12 and WebRTC: not acceptable here

the right direction? Below is my configuration. The sofpthone is registered as 1060. Thanks in advance. Marco Signorini. pjsip.conf: [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/etc/asterisk/sslcert.pem method=tlsv1 [1060] type=endpoint transport=transport-tls context=from-i

Re: [asterisk-users] WSS over Asterisk

where the SIPML5 seems not able to connect to the asterisk box. Thank you and best regards, Marco Signorini. On 06/12/2014 03:21 AM, Steve Ng wrote: I am using Asterisk v12.3. As far as DTLS, I understand that applying the following Javascript will temporarily fix for SIPML5 to Asteri

[asterisk-users] R: Asterisk and Call Hold

risk 11, but there is the same problem. I've already read all the information about canreinvite and directmedia Can anybody help me? Thanks a lot Marco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

[asterisk-users] Debugging issues with setup

Hello, I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a previous server. I have replicated all the configurations, modules and setup that I know of. However, when I tested an outbound call, it didn’t work. Checking the asterisk message log yielded nothing. Any idea

[asterisk-users] Asterisk 13, PJSIP and T38 problem

t38_udptl=yes t38_udptl_ec=fec t38_udptl_maxdatagram=400 [trunk-patton] type=auth auth_type=userpass password=X username=X = Thanks Marco -- _ -- Bandwidth and Colocation Pr

[asterisk-users] Can not calculate far_max_ifp before far_max_datagram has been set

Marco-- _ -- 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

[asterisk-users] Originate on AMI

mmand just dial both numbers using the dialplan ?? Am I too far here ? or is this something that already exists and I don't know it ?? I would appreciate any help. Thanks a lot, ______ Marco Eduardo Cordeiro Visioncom IT &

[asterisk-users] RES: How can I Disable call-waiting

Have you tried incominglimit=1 on sip.conf ?? It worked for me, no matter which softphone or ipphone / ATA I use, it works. You have to use it inside the configuration for every sip peer, just like this: [1002] Type=friend Host = dynamic Port = 5060 incominglimit=1 . . . De: [EMAIL

[asterisk-users] RES: RES: How can I Disable call-waiting

cial Discussion Assunto: Re: [asterisk-users] RES: How can I Disable call-waiting Hello thank u for ur attention but I did it and in fact its the same as call-limit in newer versions. this cmd limit ur call not disable call-waiting. best regards On Wed, Jul 23, 2008 at 5:02 PM, Marco Eduardo Cord

[asterisk-users] RES: a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal)

Hello, Just wanted to let you know that the XP version works fine on vista. I was working on a similar program but didn’t have enough time to finish, I was working on Delphi 7 btw. Thanks Marco. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Gerald

[asterisk-users] RES: Queue Penalties not working properly

You have to limit calls to these agents, use incomminglimit or call-limit on sip.conf to do that. That way, when the first agent answers a call, all the other calls directed to it will return with busy signal, and will be transferred to the other agent. __ Marco

[asterisk-users] SIP Trunk - problem to connect

Hello! Thnxs for reading! I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset, for instance (and it works!) Connection parameters are: Authentication Name: Número 11 Authentication password: 12345678 Username: 11 Display name: 11 Domain: hpbx.iplanne

[asterisk-users] Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone

or helping me gettng started with asterisk Marco -- _ -- 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

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