[Asterisk-Users] Re: How can I silently use ASTCC?

Hope this helps. -Barry Flanagan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: ASTCC questions

this helps. -- -Barry Flanagan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ASTCC and CDR info

info? -- -Barry Flanagan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] re: webvmail

PROTECTED] i.e. [EMAIL PROTECTED] or whatever. You could edit vmail.cgi to change the default context. Look for the line: if (!$context) { $context = default; } and change default to the default context you want it to be. Regards, -- -Barry Flanagan

RE: [Asterisk-Users] Ser and Asterisk together

, 5060); } -- -Barry Flanagan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Ser and Asterisk together

ser.cfg file? Thanks. -- -Barry Flanagan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Question on SIP Domains and registration

, and this means that I have clashes. I need to find a way that asterisk can differentiate between [EMAIL PROTECTED] and [EMAIL PROTECTED] Does anyone have any idea? Asterisk is being used as the PSTN gateway, where all registrations take place on OpenSER. Thanks. -- -Barry Flanagan

[Asterisk-Users] SIP domain support for authentication and virtual hosting

Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] Thanks.. -- -Barry Flanagan

[Asterisk-Users] Question on SIP authentication with users from OpenSER

have seen instances where if I have a static SIP entry with the same host= line, a non-existent user will be accepted as this static user. 3. How can have more than one possible host= setting for a user (i.e. they could come in from either of our OpenSER servers. Thanks! -- -Barry

[Asterisk-Users] Re: How do I install speex for asterisk?

: disallow=all allow=ulaw allow=alaw allow=gsm allow=speex Hope this helps. -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Re: Fromstring when sending e-mail on recieved voicemail

want to change both sender-address and the name of the sender. This is actually picked up out of /etc/passwd by default AFAIK. In voicemail.conf you can change the serveremail and fromstring settings, although I think this is only in 1.2.x. Hope this helps. -Barry Flanagan

[Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail

the From setting. Your server is running as the user asterisk and if you add that to the trusted list it might do the trick. Sorry, but I don't know what sendmail Gentoo uses so can't tell you exactly how to do this. Hope this helps. -Barry Flanagan -Opprinnelig melding- Fra

[Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail

, 'usermod -c New Asterisk Description asterisk' should do the trick! -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail

Arne Morten Johansen wrote: As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not trusting the asterisk user. And the default behaviour of that is to not allow modification of the fromstring and serveremail. So if you have any idea how to fix that

[Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail

to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant ; -Barry Flanagan -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan Sendt: 22

[Asterisk-Users] Re: SV: Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail

arguments can be found on the ; strftime(3) man page ; ; Default emaildateformat=%A, %B %d, %Y at %r ; 24h date format ;emaildateformat=%A, %d %B %Y at %H:%M:%S Hope this helps. -Barry Flanagan -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry

[Asterisk-Users] Re: context being ignored by inbound sip call

, just knowing the username. Setting it as friend works, but my feeling is (and your comment above re-enforces this) that this is not the way to do it. How do I set up so that the host MUST match one of our OpenSER IPs AND the username must match? Thanks. -- -Barry Flanagan

[Asterisk-Users] Re: a2billing without IVR

to the user after the call (values : yes - no) say_balance_after_call=NO ; Play the time the user can call (values : yes - no) say_timetocall=NO Hope this helps. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Re: a2billing without IVR

:[EMAIL PROTECTED] wrote: On Fri, 2006-02-24 at 10:58 +, Barry Flanagan wrote: Asterisk Sales wrote: mailto: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hello list, Is there any way to use a2billing without the IVR

[Asterisk-Users] Problem with uac_replace and corrupted From

to openser in a month. The only thing I did was upgrade Asterisk to 1.2.5 Anyone any ideas? -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Attended Transfer - transfer timeout, how to change?

changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit

[Asterisk-Users] Attended transfers timing out after 3 rings

. This is a big problem for us, so any help is much appreciated! -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Attended Transfer - transfer timeout, how to change?

Kevin Bockman wrote: Barry Flanagan wrote: Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies

[Asterisk-Users] Problem with intermittent one-way audio

18 not within window 19-19 Any help much appreciated. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] Problem with intermittent one-way audio

Doug Lytle wrote: Barry Flanagan wrote: Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where

Re: [Asterisk-Users] VoIP prepaid billing

://www.asterisk2billing.org/) and find it works well. You will find others at http://www.voip-info.org/, along with a wealth of other useful Asterisk information. Hope this helps. -- -Barry Flanagan ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] Problem with intermittent one-way audio

Peter Fern wrote: I've had the same problem with all boxen running the same version. We ditched IAX2 for SIP and it has been working fine since. Well, upgrading my remote site to 1.2.5 appears to have fixed my issues. -Barry Doug Lytle wrote: Barry Flanagan wrote: Hi, I have

Re: [Asterisk-Users] Realtime SIP Persistency

, and MWI does not happen. I think though if you use rtcachefriends=yes in your [general] section of sip.conf that it will work as you desire. Hope this helps. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] VoIP prepaid billing

it to drop back to the main IVR and still monitor outgoing time? Not that I know of. -Barry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Flanagan Sent: Tuesday, March 21, 2006 2:00 PM To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] How to hide CallerID - SetCallerPres(prohib) not working

else I need to make this work? I can't just set the CallerIDNUM to null, as it is needed for billing purposes. Thanks. -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

Re: [Asterisk-Users] How to hide CallerID - SetCallerPres(prohib) not working

Barry Flanagan wrote: Hi, Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on certain extensions. I have usecallingpres=yes in zapata.conf, and am using SetCallerPres(prohib) in my dialplan prior to the Dial command. No matter what I set SetCallerPres to the CID is still

Re: [Asterisk-Users] Database server

to ensure that the sql driver from asterisk-addons was compiled using the libraries from the new version of MySQL. Make sure you have the new mysql development packages installed and then re-compile asterisk-addons. Hope this helps. -- -Barry Flanagan

Re: [Asterisk-Users] Anyone using VoIP WiFi phones?

DECT handsets cost less than 50 EURO. The ATA also takes in my landline, so I only have one set of phones for both. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

[Asterisk-Users] Impport script for upgrading to 1.2 SQL Realtime?

Hi, Is there a script anywhere which would import existing *.conf entries into a mysql database for use with the realtime architecture? Thanks in advance. -- -Barry Flanagan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk

[Asterisk-Users] SIP domain support for authentication and virtual hosting

Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] Thanks.. -- -Barry Flanagan

[asterisk-users] Possible Security issue with Kamailio - Asterisk Realtime integration

an exact username match? Thanks -Barry Flanagan -- _ -- 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

Re: [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition

On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten = s,n,Dial(SIP/peer1/number@**domain2.com num...@domain2.com ,60,r) [peer1] type=friend

Re: [asterisk-users] Queue Limit Callers

difference to our avg hold time, as queues are no longer competing against one anther for available agents. Hope this helps. -Barry Flanagan I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls

Re: [asterisk-users] Queue Ring inuse is shared ?

On 22 June 2013 10:11, Shanavaz E A shanava...@yahoo.com wrote: Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. Should that not be ringinuse = no? -Barry --

Re: [asterisk-users] Add SIP Header for 1 SIP peer when calling a group of SIP peers

You should be able to do this by using a Local channel for the peer you want to add the header to: exten = _,1,Dial(Local/inno0SIP/inno4SIP/inno6,30) exten = inno0,1,SipAddHeader(X-YourHeader) exten = inno0,2,Dial(SIP/inno0) Hope this helps. -Barry Flanagan Kind regards, Jonas

Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?

On Friday 14 Feb 2014, Tiago Geada wrote: Hi all, How does one detect the 'divert' to voicemail? Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones. How can asterisk know if the call is being diverted?? It can't. But you know (from the STD code)

Re: [asterisk-users] Asterisk 1.6

for '194.100.46.132 194.100.46.132:56714' - Wrong password sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Check out fail2ban. Works well. Hope this helps. -Barry Flanagan Thanks

Re: [asterisk-users] VPN SIP Phone | PC Traffic

if it extends to the PC port or not. Nice feature I agree. http://www.yealink.com/Upload/T2X/20131125/OpenVPN_Feature_on_Yealink_IP_Phones.pdf Hope this helps. -Barry Flanagan -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

of the INVITE's sip headers or body? I tried a several variations, but nothing quite worked. Something like: [peer_inbound] context=peercontext type=peer host=192.168.1.1 ...should do the job. Hope this helps. -Barry Flanagan

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

to be in the Kamailio trusted table, or explicitly test for the src_ip rather than use allow_trusted(). Hope this helps. -Barry Flanagan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Sippeers realtime with minimum table

registration details. Hope this helps. -Barry Flanagan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] compiling dahdi and exporting it to another system

it should be possible to compile it for an alternate kernel if you have the headers, but have never had to do that so) Hope this helps. -Barry Flanagan Thank you, Anthony. -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Kamallio registration

though. Hope this helps. -Barry Flanagan -- _ -- 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

Re: [asterisk-users] asterisk server stress test

seek. http://sipp.sourceforge.net/ Hope this helps. -Barry Flanagan -- _ -- 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] subscriber state before dial

> So if "2" is not registered (or is busy) then Dial(SIP/1/5/3). > > Perhaps CHANISAVAIL will do what you need: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail Hope this helps. -Barry Flanagan -- ___

Re: [asterisk-users] Asterisk 13 Realtime Voicemail frustrating issue

users default". And whet i try to access a > mailbox, i get a "Invalid password". > > Any hints ? Please, i'm really frustrated ! > I think the mailbox field in MySQL needs to be in the form 'mailbox@context' in Asterisk 13. As for your zones, these are usually defined in

Re: [asterisk-users] OpenSIPS or Kamailio based fronting for Asterisk?

a realtime/mysql DB used by the PBXs. > > Please also let me know if I should have to change my Asterisk PBX config > in any way for this to happen. > > Hi This might be a good starting point: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-aster

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

; isolated to you. If you follow the instructions on the wiki[1] it will > provide a backtrace which will show where chan_sip is hanging. > > Joshua, This was actually reported in https://issues.asterisk.org/jira/browse/ASTERISK-25468 with backtraces. It appeared to have started in 13.5, as I teste

Re: [asterisk-users] Any reason Asterisk won't start without a rebuild on a cloned VPS?

On 29 November 2016 at 11:19, Joshua Colp <jc...@digium.com> wrote: > On Tue, Nov 29, 2016, at 07:15 AM, Barry Flanagan wrote: > > On 29 November 2016 at 10:56, Jonathan H <lardconce...@gmail.com> wrote: > > > > > Any ideas why a VPS, cloned from another in

Re: [asterisk-users] Any reason Asterisk won't start without a rebuild on a cloned VPS?

version etc. > You probably need to select "DONT_OPTIMIZE" in make menuselect under "Compiler Flags". This is generally required for VPS or any situation where the binary might be used on a slight

Re: [asterisk-users] OT: Want to capture all SIP messages

On 31 May 2017 at 21:29, Barry Flanagan <barryf-li...@flanagan.ie> wrote: > Voipmonitor, or sngrep > > Sorry, didn't see the "long term" bit. voipmonitor or Homer are your best best. -Barry Flanagan > -Barry Flanagan > > > > > > >

Re: [asterisk-users] OT: Want to capture all SIP messages

Voipmonitor, or sngrep -Barry Flanagan On 31 May 2017 at 20:36, Steve Edwards <asterisk@sedwards.com> wrote: > I want to capture all SIP messages. > > I have about 30 hosts in about 6 colos. > > My first thought was dumpcap, but the output file name format bugs me

Re: [asterisk-users] softphone instead of desktop phones

On 30 April 2017 at 16:54, Tech Support wrote: > I thought this was a non-commercial list. > > Yeah, I wouldn't mind so much if it had actually answered the original poster's query. "Switch to our proprietary solution and we can offer you this proprietary solution"

Re: [asterisk-users] How to properly execute rasterisk over SSH ?

; The above command "rougly works" but some non-printable characters cause > undesirable issue during CLI session. > > Maybe try ssh -t root@foobar rasterisk Hope this helps. -Barry Flanagan Any suggestion ? > > Regards > -- > __

Re: [asterisk-users] How to implement an ENUM mock database ?

w.powerdns.com/) in production and it works well. Choice of backends - mysql, sqlite3, bind, etc. Works well. -Barry Flanagan Best regards > > [1] https://github.com/hadrienk/enum-dns > -- > _ > -- Bandwidth and Col

Re: [asterisk-users] Pass through registration / proxy

o what you want - no need for Asterisk. At best, you would be using Asterisk as a b2bua between your endpoint(s) and legacy PBX. -Barry Flanagan > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *Carlos Rojas > *Sent:

Re: [asterisk-users] Half Off Topic Questions

On 6 March 2018 at 09:49, Antony Stone wrote: > On Tuesday 06 March 2018 at 09:05:25, Markus Weiler wrote: > > > Hi Group, > > > > we're just wondering, in German we call the different types of > phone-numbers > > (Geographic,mobile,national,VoIP...)