[asterisk-users] Dahdi Dropping Calls

Hi Guys Has anyone seen Dahdi dropping incoming calls with Hangup cause 27? It only drops whilst we are on the phone? Its not every single call Any ideas? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony

[asterisk-users] Centos 6 mISDN

Hi Guys Has anyone got this working on Centos 6? -- _ -- 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] basic sip quesiton

Put disallow=all below all of the allow= -- _ -- 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] My digium card die?

Can you confirm Dahdi is loaded correctly What does the output of dmesg show? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of PedroTron Sent: 02 September 2012 04:28 PM To: Asterisk Users Mailing List -

Re: [asterisk-users] Dahdi Answer a Call On ringing State.

More info??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Parveen Lamba Sent: 13 September 2012 01:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing

Re: [asterisk-users] B200p card - use dahdi or mISDN?

I have worked with the B200P before and used the standard mISDN and the standard DAHDI and both worked fine. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists Sent: 16 October 2012 12:30 PM To:

Re: [asterisk-users] dahdi driver not getting install

Do a yum install kernel-devel kernel-headers Reboot and it will work Sent from my iPhone On 11 May 2013, at 12:20 PM, Alec Davis siva...@paradise.net.nz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On

Re: [asterisk-users] dahdi driver not getting install

I thought he said rhel 6.3 Sent from my iPhone On 11 May 2013, at 2:48 PM, Asghar Mohammad asghar...@gmail.com wrote: he is using debian. debian have yum? On Sat, May 11, 2013 at 2:44 PM, Andrew Colin and...@vsave.co.za wrote: Do a yum install kernel-devel kernel-headers Reboot

[asterisk-users] Error 488 Not Acceptable Here

Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] asterisk ip authentication

Andrew Colin Technical Director T:010 591 4358 C: 082 310 3007 and...@vsave.co.za On 7/28/2013 6:26 PM, james jan wrote: allow=g729 allow=alaw allow=ulaw allow=gsm

Re: [asterisk-users] asterisk ip authentication

I just find it insecure because if someone does hack they can use any codec. I suppose not very insecure but I like to lock things down as much as possible. On 7/28/2013 9:09 PM, Matt Behrens wrote: On Jul 28, 2013, at 2:59 PM, Andrew Colin and...@vsave.co.za wrote: if you say allow=all

Re: [asterisk-users] asterisk ip authentication

send me a copy of your sip config also make sure dissallow is before allow. Kind Regards Andrew Colin Technical Director T:010 591 4358 C: 082 310 3007

Re: [asterisk-users] asterisk ip authentication

remove disallow completely you are basically saying do not allow anything then allow anything so remove the disallow part and leave allow Kind Regards Andrew Colin

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

change server two to host = dynamic then add register = on server 1 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N gopalakrishnan...@gmail.com mailto:gopalakrishnan...@gmail.com wrote:

Re: [asterisk-users] RTP port ranges

Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. -- _ -- Bandwidth and

Re: [asterisk-users] RTP port ranges

Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall

[asterisk-users] Strange Error

Hi Guys, Anyone ever seen this before. on asterisk 1.8 if i set one of my pabx extensions to show private number and send a call over VoIP with g729 the call fails but with alaw it works. If i enable the callerid on g729 it also works see error below From:

Re: [asterisk-users] Asterisk uses 105% CPU

Are you transcoding? What is your server spec? Regards Andrew Colin-mobile Vsave(PTY)Ltd Original message From: Jonas Kellens jonas.kell...@telenet.be Date:27/11/2013 13:48 (GMT+02:00) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users

Re: [asterisk-users] stopping unwanted attempts

Fail2ban works well otherwise you can write your own script im bash or perl to block them in iptables Regards Andrew Colin-mobile Vsave(PTY)Ltd Original message From: Jerry Geis ge...@pagestation.com Date:18/01/2014 10:59 PM (GMT+02:00) To: asterisk-users

Re: [asterisk-users] stopping unwanted attempts

Geoip works well to block all countries except your own Regards Andrew Colin-mobile Vsave(PTY)Ltd Original message From: Eric Wieling ewiel...@nyigc.com Date:19/01/2014 8:40 PM (GMT+02:00) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users

Re: [asterisk-users] Attack on Sip server.

Block the ip? You should only enable sip for your specific clients in iptables. Sent from Samsung Mobile div Original message /divdivFrom: arun kumar arunvsadni...@gmail.com /divdivDate:27/06/2014 4:42 PM (GMT+02:00) /divdivTo: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] recording in mp3

Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile div Original message /divdivFrom: Sameer Rathod sam...@hostnsoft.com

Re: [asterisk-users] recording in mp3

and directly writing MP3 files. On Mon, Jun 30, 2014 at 3:11 PM, andrew Colin and...@vsave.co.za wrote: Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung

Re: [asterisk-users] recording in mp3

: Re: [asterisk-users] recording in mp3 /divdiv /divwhat is your interface? On 1/7/2014 19:13, andrew Colin wrote: Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next

[asterisk-users] Call rating software

Hi Guys Does anyone know of any good cdr rating software. I am looking for something that I can pull reports by extension.  Not a full billing solution like a2billing. Sent from Samsung Mobile-- _ -- Bandwidth and Colocation

Re: [asterisk-users] Call rating software

List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call rating software On Tuesday 01 Jul 2014, andrew Colin wrote: Hi Guys Does anyone know of any good cdr rating software. I am looking for something that I can pull reports by extension. Not a full billing solution like

Re: [asterisk-users] Call rating software

Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] Call rating software /divdiv /divOn Wednesday 02 Jul 2014, Andrew Colin wrote: Can you try maybe assist with this, as I have tried for ages and still cant get it right. Firstly, have you got

[asterisk-users] Strange Error

Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] recording in mp3

Can you explain? Sent from Samsung Mobile div Original message /divdivFrom: Tiago Geada tiago.ge...@gmail.com /divdivDate:03/07/2014 9:04 PM (GMT+02:00) /divdivTo: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /divdivSubject: Re:

Re: [asterisk-users] Realtime ERROR

Hi Guys, I have recently moved my database servers to a different database cluster that runs on haproxy. Every minute or so I get the following error in asterisk MySQL RealTime: Ping failed (2006). Trying an explicit reconnect The strange thing is if I do realtime mysql status It

Re: [asterisk-users] Realtime ERROR

Hi Rainer, I am using roundrobin From: Rainer Piper [mailto:rainer.pi...@soho-piper.de] Sent: Thursday, September 25, 2014 6:21 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime ERROR Am 25.09.2014 um 16:24 schrieb

[asterisk-users] One way audio internal

Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: mailto:and...@convergedgroup.net and...@convergedgroup.net Web: http

Re: [asterisk-users] One way audio internal

I am using the free g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http

Re: [asterisk-users] One way audio internal

I currently am running on a Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz Codec im using is codec_g729-ast18-icc-glibc-x86_64-core2.so -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] One way audio internal

I currently am running on a Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz Codec im using is codec_g729-ast18-icc-glibc-x86_64-core2.so Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607

Re: [asterisk-users] BlindXfer Sensitivity

/divdivFrom: Kevin Larsen kevin.lar...@pioneerballoon.com /divdivDate:16/02/2015 17:11 (GMT+02:00) /divdivTo: Andrew Colin and...@convergedgroup.net,Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] BlindXfer

Re: [asterisk-users] BlindXfer Sensitivity

RFC2833 The strange thing is how asterisk is not registering she has pushed ## on those Rare occiasions On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin and...@vsave.co.za wrote: The strange thing is its only sometimes my dial string is as follows exten = s,1, Dial (SIP/200,, tT

[asterisk-users] BlindXfer Sensitivity

Hi Guys We have a client running on a polycom vvx400 IP phone on our asterisk 1.8.18 system The issue we have is the switchboard lady uses ## to transfer calls but sometimes it just does not work and just plays the DTMF tone to the calling party. Is there any way to adjust the

Re: [asterisk-users] queue reload command

Hi queue reload(queue name) or queue reload all for example queue reload reception From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, January 8, 2015 2:10 PM To: Asterisk Users Mailing List -

[asterisk-users] 4 Port PRI

Hi Guys I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en

Re: [asterisk-users] 4 Port PRI

4 Port PRI sangoma a104 From: jg [mailto:webaccounts...@jgoettgens.de] Sent: Wednesday, March 18, 2015 2:09 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port

[asterisk-users] Strange Polycom Issue

Hi Guys, We are getting a strange issue on certain polycom phones, sometimes when a call comes in it just flashes to show there is a call but does not play any sound. This problem is very intermittent and happens to maybe 2 out of 10 calls. Has any else experienced this before? --

[asterisk-users] Yealink t26 and T28 Panels

Hi Guys We have a strange a strange issue at a client they have 3 panels on their phone and every so often the panels reboot themselves yet the phone stays on. We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact same issue. Has anyone seen this

Re: [asterisk-users] Yealink t26 and T28 Panels

Originally we used just POE but now each of the 3 panels has its own PSU From: jg [mailto:webaccounts...@jgoettgens.de] Sent: Friday, March 13, 2015 11:18 AM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yealink t26 and T28 Panels

[asterisk-users] Kamallio registration

Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? --

[asterisk-users] Delayed RTP

Hi Guys We have a strange issue whereby one phone has delayed rtp So what happens is when the lady answers the phone for the 1st 1 second they can not hear her and then everything is fine I am running asterisk 1.8.28.0 Has anyone seen this before? --

Re: [asterisk-users] Product CDR/Queue/Meetme

Hi Helvio I will be interested to test your product and give you some feedback. . Sent from my Samsung Galaxy s6 smartphone. Original message From: Helvio Junior helvio.lis...@gmail.com Date: 29/06/2015 20:58 (GMT+02:00) To: Abdul Basit basit.e...@gmail.com, Asterisk

[asterisk-users] Prodding channel Failed

Hi Guys I am seeing this error a lot in the CLI lately What does it mean? Prodding channel SIP/XXX failed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] Modify Contact in PJsip

Colp <jc...@digium.com> Date: 2015/10/19 13:03 (GMT+02:00) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modify Contact in PJsip On 15-10-19 07:41 AM, Andrew Colin wrote: > Hi Guys > > We are using the wizard to configure our pjsip trunk(see below) &g

Re: [asterisk-users] Modify Contact in PJsip

Do you know if this can be achieved with the standard sip stack in asterisk? Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10 591 4600 Email:  and...@convergedgroup.net Web:  http

Re: [asterisk-users] Modify Contact in PJsip

Ok thanks Joshua Do you know what this error means when I dial out in pjsip and the call fails Unable to create request with auth.No auth credent als for any realms in challenge Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258

[asterisk-users] Modify Contact in PJsip

Hi Guys We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk [trunk_defaults](!) type = wizard transport = transport-udp

[asterisk-users] PJSIP Dialout error

Hi Guys I keep getting this "Warning" when I dial out via pjsip and the calls fail But if I do a pjsip reload it works for 1 minute WARNING[6707]: res_pjsip_outbound_authenticator_digest.c:135 digest_create_request_with_auth_from_old: Unable to create request with auth.No auth credentials

[asterisk-users] Change Contact field in sip invite

Hi Guys Does anyone know of a way I can change the contact field in the sip invite to display sip:username:ip instead of sip:did:ip We need to do this without changing the from field. I tried using fromuser=username but that just modifies both the contact and the from parameter I know

Re: [asterisk-users] Storing HANGUPCAUSE in CDR

You can use this exten => h,1,Set(CDR(userfield)=Hangupcause:${HANGUPCAUSE}) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ross Beer Sent: Friday, October 9, 2015 1:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] Call Return

Hi Guys I am trying to write a macro for a call return so for example Anyone in the company transfers a call to another extension and it is not answered etc it must return to the person who did the transfer I have got it working but if the call originates externally for example someone calls

Re: [asterisk-users] Call Return

Hi Aj Can you perhaps show me an example as to how you would do it as I have tried setting it very early but still doesn’t work Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10 591 4600 Email

Re: [asterisk-users] Calls are dropped after 15 minutes

I had a similar issue and i set a timeout which fixed the issue SIP/trunk/ ${EXTEN},216,t We only had this on one of our providers the rest we havent had the issue - Original Message - From: Steve Edwards To: Asterisk Users Mailing List - Non-Commercial