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 Discussion
Sent: Sat, 30 Jul 2016 20:27:45 +0200 (SAST)
Subject: Re: [asterisk-users] Calls are dropped after 15 minutes
On Sat, 30 Jul 2016, Keith Heppner wrote:
> We have a problem in that calls are dropped after 15 minutes (on both
> internal and out going calls, incoming calls do not seem to have that
> limit) How do we fix it?
You may gain some insight from viewing the console output after bumping up
the debug and verbose levels.
You will probably resolve this by using tcpdump to capture packets and
wireshark to see what's happening.
I had a problem with a similar description that was resolved by refusing
SIP session timers.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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[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 live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modify Contact in PJsip
Hi Joshua If i put the default_user option per endpoint would it work? So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We basically need to achieve the same functionality Thanks Original message From: Joshua 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) > > 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 > The Contact header can not currently be modified on a per-endpoint basis and takes its values from the generated From header. On a global scale it could be controlled using the default_user global option. Otherwise there's no real way without adding explicit support for it. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you. E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Monday, October 19, 2015 2:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modify Contact in PJsip On 15-10-19 08:17 AM, Andrew Colin wrote: > Hi Joshua > > If i put the default_user option per endpoint would it work? No, it's a global only option. > > So what exactly does the contact_user option do? It sets the Contact user in an outbound registration so that the URI dialed by the remote SIP server may contain that user (or may not, depending on their configuration/deployment). > > I know that in freeswitch there is the option extension-in-contact. > We basically need to achieve the same functionality It would require modifying the code and adding support. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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/IECNS/JAN/09) Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you. E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Monday, October 19, 2015 2:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modify Contact in PJsip On 15-10-19 09:12 AM, Andrew Colin wrote: > Do you know if this can be achieved with the standard sip stack in asterisk? If you are referring to chan_sip I don't believe so but it is possible there is some obscure option or method to do it that I am aware of. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30 registration/expiration = 1800 contact_pattern=xxx [xxx](trunk_defaults) sends_auth = yes sends_registrations = yes endpoint/context = extensions remote_hosts = xxx.xx.xx.xx accepts_registrations = no endpoint/send_rpid = yes endpoint/send_pai = yes outbound_auth/username = xxx outbound_auth/password = xxx contact_pattern=xxx -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 for any realms in challenge. Any ideas? -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR
This was always possible in the past, however does not work in the current
release.
I believe this is a bug.
_
To: asterisk-users@lists.digium.com
From: cerv...@fpf.slu.cz
Date: Fri, 9 Oct 2015 10:04:47 +0200
Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR
search in archives
save the records to another table like cdr_extended
Dne 7.10.2015 v 15:26 Ross Beer napsal(a):
Hi,
I have the following code that operates when a channel is hung-up:
[record-hangupcause]
exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})
exten => s,n,Return()
Before the dial a hangup handler is registered:
Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1)
The routine is called and the variables are being set, however not on the
channel's CDR which made the call. I believe this is due to the CDR being
closes as soon as the dial returns.
By changing the cdr option 'endbeforehexten=no' this should keep the CDR
accessible, however all this does is cause another CDR to be created for the
'h' extension.
Is there a way to update the CDR so that a result can be stored per dial?
Thank you in advance,
Ross
--
---
Marek Cervenka
===
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[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 in freeswitch they use the parameter extension-In-Contact Has anyone managed to do this before? -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: and...@convergedgroup.net
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you.E-mail including attachments is susceptible to data
corruption, interception, unauthorized amendment, tampering and viruses, and
we only send and receive emails on the basis that we are not liable for any
such corruption, interception, amendment, tampering or viruses or any
consequences thereof.
-Original Message-
From: A J Stiles [mailto:asterisk_l...@earthshod.co.uk]
Sent: Thursday, July 9, 2015 10:03 AM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Return
On Wednesday 08 Jul 2015, Andrew Colin wrote:
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 in to the switchboard and they transfer it then
it tries to return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in
this case 200) and declare it as a variable?
You need to set a variable quite early in your extension logic, using a Set
command;
Set(dialled=${EXTEN})
and then later you can retrieve it as ${dialled} . This variable will
persist across context jumps, even although ${EXTEN} may have changed.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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[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 in to the switchboard and they transfer it then it tries to
return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in this
case 200) and declare it as a variable?
--
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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 Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Product CDR/Queue/Meetme 1.8 or higher. Att, Hélvio Junior SafeId - Gestão de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.jun...@safetrend.com.br On 29/06/2015 14:43, Abdul Basit wrote: Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support aster...@voipbusiness.us mailto:aster...@voipbusiness.us wrote: Please keep the “me to” emails off the list. Regards; JV *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Magno Guimarães *Sent:* Monday, June 22, 2015 3:55 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Product CDR/Queue/Meetme Hello, I am interested, too. Att, Welinghton Citando Mitul Limbani mi...@enterux.in mailto:mi...@enterux.in: Hey Helvio, Would like to check it out as well. Do email me, Mitul On 22-Jun-2015 9:05 AM, Helvio Junior helvio.lis...@gmail.com mailto:helvio.lis...@gmail.com wrote: Gentleman, Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me. I had some difficult looking for a Asterisk software that provide me some functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i decided to develop than. In a few weeks i'll deploy a Beta version of this software and i'd like to know if is somebody available to try this beta and free version? If you don't want to try this version but would like to see/suggest any feature in this software, let me know. Forecast functions to Beta Version: * Realtime view for: o Queues; o Peers (Similar as BLF); o Trunk calls/utilization; * MeetMe o Create, modify, delete and schedule; o Real time view of members; o Delete members; o Mute/Unmute; o Send Invite by e-mail (with .VCS file) * Dialer o Create dialer (by campaign with contacts) o Monitoring of campaig, calls, and status; o Time control to retry failed call o Control of day time to call (commercial time, full time, etc...) * Charts and reports: o Trunk utilization; o CDR; o Queues (Most common reports and charts, distributions, times, etc...) o Export to Excel Spreadsheet and PDF File o Report Scheduler o Much more... * REST API for 100% of functionalities; * Admin and User Console 100% Web HTML5; * Developed in Windows with C#; * Integrate with Asterisk using AMI only; * Allow manage many Asterisk that you want using same instance of this software (One software and one installation); Obs.: I'll provide a Full License for everybody that help me trying the Beta version. -- Att, Hélvio Junior SafeId - Gestão de identidades e Acessos +55 41 | 9893-2694,single-sign-on.com.br http://single-sign-on.com.br helvio.jun...@safetrend.com.br mailto:helvio.jun...@safetrend.com.br -- Att, Hélvio Junior SafeId - Gestão de identidades e Acessos +55 41 | 9893-2694,single-sign-on.com.br http://single-sign-on.com.br helvio.jun...@safetrend.com.br mailto:helvio.jun...@safetrend.com.br -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
[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? -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kamallio registration
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[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 switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 PRI or BRI? Which card are you using? Typically the installation script or procedure lets you configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels. jg -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 before? -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Hi! 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 before? I frequently use the newer T48G and T46G phones with the EXP40 expansion module. There are issues, if you are logged into the phone via the webinterface as an admin. Among other things, the display is not properly updated and wrong numbers may get dialed. Some time ago, there was a firmware update and I am not aware of any stability issues at the moment. How do you supply power? 3 expansion modules + the phone and a cheap POE switch could be critical. It may not be the power itself, but the correct handling of energy saving states. jg -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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? -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BlindXfer Sensitivity
The strange thing is its only sometimes my dial string is as follows exten = s,1, Dial (SIP/200,, tT) For that particular route but obviously s,3 as have Ringing () first etc. After she pushes ## 6 times it will go thru sometimes. Sent from Samsung Mobile div Original message /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 Sensitivity /divdiv /div 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 sensitivity of the blindxfer feature? The polycom Transfer button is useless as there is a big delay until it apprears I would greatly appreciate any advice It seems weird that this would be some kind of sensitivity to the DTMF tones. The first thing I would look for is on a call that she cannot blind transfer, check how the Dial command was used to reach her. Does it have the proper use of the tT options (depending on whether she called them or they called her)? I would almost bet there is a call path that occurs which doesn't have the proper options set to allow the transfer.-- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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) For that particular route but obviously s,3 as have Ringing () first etc. After she pushes ## 6 times it will go thru sometimes. How is the DTMF being transmitted from the phone to Asterisk? RFC2833, in-band, SIP INFO...? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 sensitivity of the blindxfer feature? The polycom Transfer button is useless as there is a big delay until it apprears I would greatly appreciate any advice -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 - Non-Commercial Discussion
Subject: [asterisk-users] queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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[asterisk-users] One way audio internal
Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with 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: mailto:and...@convergedgroup.net and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Andrew Colin Subject: Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 1:04 PM To: Andrew Colin Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] One way audio internal Then something to do with your codec selection priority. On 21-Nov-2014 4:26 PM, Andrew Colin and...@convergedgroup.net wrote: 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 tel:%2B27%20%280%2910%20591%204607 Mobile: +27 (0)82 310 3007 tel:%2B27%20%280%2982%20310%203007 Switchboard: +27 (0)10 591 4600 tel:%2B27%20%280%2910%20591%204600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Andrew Colin Subject: Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 shows as connected just the timer resets. Any idea why this is occurring? -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Andrew Colin: 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 shows as connected just the timer resets. Any idea why this is occurring? Hi Andrew, what balancing algorithm you use in haproxy.cfg ? balance source balance roundrobin or balance leastconn -- Rainer Piper Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: callto:004922897167161 +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de _ No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4765 / Virus Database: 4025/8267 - Release Date: 09/24/14 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: [asterisk-users]
recording in mp3 /divdiv
/divno need.
mixmonitor has a argument that is a script ran just as the recording is
finished.
we use this to move the file from ramfs to final destination.
you can use it to use sox and convert it...
On 2 July 2014 18:54, Dave Platt dpl...@radagast.org 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 day for today's recordings
If you're up to writing a bit of shell script, and are running
on Linux, you could automate the conversion process so that it
happens as soon as the recording is completed.
Look at the inotify system service (man section 7) and the
inotifywatch program. You can tell inotifywatch to monitor
files being written into a specific directory (or set of
directories) and output a series of events when files in this
directory are open or closed.
What you'd probably want to do, is catch the close_write
events (a file has been closed, and it had been opened in
a mode which allows it to be written). When you see a
close_write event for a recording file of the sort that
Asterisk writes, you'd check to see if it's been converted
to your desired format yet. If not, fire off a separate
task (e.g. via batch) to convert it.
Here's a very simple script I did to do something like this...
run a periodic-processing script a few seconds after files
with a specific name pattern have been touched in any way.
It's not sophisticated enough to look only for close or
close_wait events, but it should give you the idea.
#!/bin/bash
function processevents () {
action=0
while true ; do
if [ $action == 0 ] ; then
timeout=300
else
timeout=5
fi
read -t $timeout event
if [ $? != 0 ] ; then
action=0
/data/soundchaser/periodic
else
if [[ $event =~ .wav || $event =~ .gotit ]] ; then
action=1
fi
fi
done
}
cd /data/soundchaser
inotifywait -m /data/soundchaser/public_html/done | processevents
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Re: [asterisk-users] Call rating software
Can you try maybe assist with this, as I have tried for ages and still cant get it right. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: 02 July 2014 11:14 AM To: Asterisk Users Mailing 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 a2billing. Have you thought of rolling your own? It's not hard to write a program in Your Favourite Scripting Language™ to pull the records you want from the database and create a CSV spreadsheet. And then have it started by cron, to give you regular automatically-generated reports; either e-mailed directly to you, or downloadable via a web page. Alternatively, you could generate your spreadsheet as an .fods (flat XML) file. This is slightly more effort; but it has the advantage of supporting styling of table cells, so your report can be made to look pretty. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4592 / Virus Database: 3986/7781 - Release Date: 07/02/14 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call rating software
Currently I am writing to mysql With all the default fields in the cdr table in the asterisk databas Sent from Samsung Mobile div Original message /divdivFrom: A J Stiles asterisk_l...@earthshod.co.uk /divdivDate:02/07/2014 5:50 PM (GMT+02:00) /divdivTo: Asterisk Users 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 CDR working and writing to some sort of database? We use cdr_mysql; although the more modern recommendation is to use cdr_odbc (which is more generic, and will work with various database types) even if you are using a MySQL database. If you haven't got your CDR going into a database, then you need to sort that out *first*. Once you have CDR working, then it's simply a question of determining what SQL queries you need to generate to produce your report; then writing a program to build up the queries, extract the results and present them in the form you want. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recording in mp3
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 day for today's recordings Sent from Samsung Mobile div Original message /divdivFrom: binary dreamer.bin...@gmail.com /divdivDate:01/07/2014 6:09 PM (GMT+02:00) /divdivTo: asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] recording in mp3 /divdiv /divi would go for recording into wav. then at regular intervals eg every night at 01:00 i would start a script to convert the wav to mp3 and then delete the wav files. it is really easy. On 30/6/2014 23:30, Scott Griepentrog wrote: You will not be able to able to save much space if any by using MP3 instead of ulaw or wav -- at least not without expending a lot of CPU time to encode the file at a very low bitrate which sounds pretty bad even with just speech. One of the better space savings options for recordings or voicemail is gsm. Of course, using an MP3 format just because you prefer that is understandable. Additionally, I'm nearly 100% certain that Asterisk does not support encoding 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 Mobile Original message From: Sameer Rathod Date:30/06/2014 9:23 PM (GMT+02:00) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fwd: Regarding packet2packet bridging Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recording in mp3
Currently using tikal crystal call recording Do you guys know of any better ones? Sent from Samsung Mobile div Original message /divdivFrom: binary dreamer.bin...@gmail.com /divdivDate:01/07/2014 6:33 PM (GMT+02:00) /divdivTo: asterisk-users@lists.digium.com /divdivSubject: 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 day for today's recordings Sent from Samsung Mobile Original message From: binary Date:01/07/2014 6:09 PM (GMT+02:00) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] recording in mp3 i would go for recording into wav. then at regular intervals eg every night at 01:00 i would start a script to convert the wav to mp3 and then delete the wav files. it is really easy. On 30/6/2014 23:30, Scott Griepentrog wrote: You will not be able to able to save much space if any by using MP3 instead of ulaw or wav -- at least not without expending a lot of CPU time to encode the file at a very low bitrate which sounds pretty bad even with just speech. One of the better space savings options for recordings or voicemail is gsm. Of course, using an MP3 format just because you prefer that is understandable. Additionally, I'm nearly 100% certain that Asterisk does not support encoding 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 Mobile Original message From: Sameer Rathod Date:30/06/2014 9:23 PM (GMT+02:00) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fwd: Regarding packet2packet bridging Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 /divdivDate:30/06/2014 9:23 PM (GMT+02:00) /divdivTo: asterisk-users@lists.digium.com /divdivSubject: [asterisk-users] Fwd: Regarding packet2packet bridging /divdiv /div Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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 Discussion asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] Attack on Sip server. /divdiv /divHi, Change the protocol from tcp to udp in iptables. ~Arun On 27 Jun 2014 20:07, Anurag Rana anuragrana31...@gmail.com wrote: Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the packets from attacker contains string 'VaxSIPUserAgent/3.1' ) but it failed. Packets are still flowing in. iptables -I INPUT 1 -p tcp --dport 5060 -m string --string VaxSIPUserAgent --algo bm -j DROP Its something like this Registration from '30 sp:30@my_public_ip:5060 failed for '192.168.xxx.xxx:6373' - Wrong Password and there are approx 10 request per minute of this type. Please suggest some way to stop this. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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@lists.digium.com Subject: Re: [asterisk-users] stopping unwanted attempts It is far worse when you have multiple phones behind the same public address (i.e. NAT). If any one of the phones has a bad password and the IP gets blocked by fail2ban, then all phones at that site would be blocked. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Sunday, January 19, 2014 10:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] stopping unwanted attempts On 19/1/14 2:57 pm, Ron Wheeler wrote: fail2ban is so easy to set up, there is no reason not to set it up. One of the dangers with fail2ban - at least in its default configuration - is that a legitimate SIP phone with an incorrect password can quite easily send dozens of registration attempts in a couple of minutes, thus blocking that IP. -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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@lists.digium.com Subject: [asterisk-users] stopping unwanted attempts I see MANY of these in my log files: [Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from '202 sip:202@X:5060' failed for '37.8.12.147:26832' - Wrong password [Jan 15 03:06:19] NOTICE[14129] chan_sip.c: Registration from '5001 sip:5001@X:5060' failed for '37.8.12.147:21268' - Wrong password [Jan 15 03:06:23] NOTICE[14129] chan_sip.c: Registration from '30 sip:30@X:5060' failed for '37.8.12.147:21270' - Wrong password [Jan 15 03:06:48] NOTICE[14129] chan_sip.c: Registration from '70 sip:70@X:5060' failed for '37.8.12.147:21328' - Wrong password [Jan 15 03:06:50] NOTICE[14129][C-0085] chan_sip.c: Call from '' (8.33.7.110:5103) to extension '889011972592735467' rejected because extension not found in context 'default'. [Jan 15 03:06:56] NOTICE[14129] chan_sip.c: Registration from '4 sip:4@X:5060' failed for '37.8.12.147:21272' - Wrong password [Jan 15 03:07:11] NOTICE[14129] chan_sip.c: Registration from '12001 sip:12001@X:5060' failed for '37.8.12.147:5060' - Wrong password [Jan 15 03:34:02] NOTICE[14129][C-0086] chan_sip.c: Call from '' (172.246.236.90:5078) to extension '8889011972595301123' rejected because extension not found in context 'default'. What is the correct way to block these idiots so they don't even get this far. Thanks, Jerry -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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@lists.digium.com Subject: Re: [asterisk-users] Asterisk uses 105% CPU On 27-11-13 12:26, Jonas Kellens wrote: Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init What can be causing such a high load of the asterisk proces ?? There are about 35 calls with G711a codec, no translation. Kind regards, Jonas. I want to add some more information. Maybe someone knows how to help me with this information : sip*CLI core show threads 0x7f98f87fd700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8ae5700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9229700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9609700 netconsole started at [ 1423] asterisk.c listener() 0x7f98f8971700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8ec5700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8e49700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9a65700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f97f9700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8a69700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8dcd700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8d51700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9ae1700 shaun_of_the_dead started at [ 2141] app.c ast_safe_fork() 0x7f98f9b5d700 inotify_daemon started at [ 334] stdtime/localtime.c add_notify() 0x7f98f9def700 autoservice_run started at [ 219] autoservice.c ast_autoservice_start() 0x7f98f9ee7700 monitor_sig_flags started at [ 4097] asterisk.c main() 0x7f98f9f63700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98f9fdf700 cleanup started at [ 414] pbx_realtime.c load_module() 0x7f98fa05b700 scan_thread started at [ 885] pbx_spool.c load_module() 0x7f98fa0d7700 do_monitor started at [ 4684] chan_unistim.c restart_monitor() 0x7f98fa153700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98fa1cf700 process_clearcache started at [ 2265] pbx_dundi.c start_network_thread() 0x7f98fa2c7700 network_thread started at [ 2263] pbx_dundi.c start_network_thread() 0x7f98fa24b700 process_precache started at [ 2264] pbx_dundi.c start_network_thread() 0x7f98fa343700 do_monitor started at [ 1167] chan_phone.c restart_monitor() 0x7f98fa3bf700 lock_broker started at [ 509] func_lock.c load_module() 0x7f98fa43b700 network_thread started at [12310] chan_iax2.c start_network_thread() 0x7f98fa4b7700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa533700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa5af700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa62b700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa6a7700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa723700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa79f700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa81b700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa897700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa913700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa98f700 sched_run started at [ 186] sched.c ast_sched_thread_create() 0x7f98faa0b700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98faa87700 do_monitor started at [ 3897] chan_mgcp.c restart_monitor() 0x7f98fab03700 do_monitor started at [ 6647] chan_skinny.c restart_monitor() 0x7f98fab7f700 accept_thread started at [ 7358] chan_skinny.c
[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: Anonymoussip:anonymous@anonymous.invalid;user=phone;tag=07d44838 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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:
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port
extensions.conf
[man02-trunk]
exten = _1X.,1,Dial(SIP/usman02/${EXTEN})
exten = _1X.,n,Hangup
Server2
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.10.10.81
context=us02-trunk-inbound
port=5060
qualify=yes
disallow=all
allow=g729
;allow=ulaw
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=port,invite
extensions.conf
[us02-trunk-inbound]
exten = _X.,Dial(SIP/${EXTEN},60)
Now when I dial from server1, in the server 2 am getting the error as,
[Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266
check_auth: username mismatch, have 2001, digest has usman02
things are fine.. but I dont know where the mistake is...!
Can you some one advise me... !
Thanks.
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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
and...@vsave.co.za
On 7/29/2013 1:07 AM, james jan wrote:
hi all,
i've changedallow=all
and restarted service.
butstill
gives488 Not
acceptable here
The
softswitch sends codec g729.
"core show translation" says codec g729 alsa installed.
On Sun, Jul 28, 2013 at 10:11 PM,
Andrew Colin and...@vsave.co.za
wrote:
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 it will work but thats not secure at all.
How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me.
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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 Technical Director T:010 591 4358 C: 082 310 3007 and...@vsave.co.za On 7/29/2013 9:48 AM, james jan wrote: hi Andrew, here is my sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=all On Mon, Jul 29, 2013 at 9:17 AM, Andrew Colin and...@vsave.co.za wrote: 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 and...@vsave.co.za On 7/29/2013 1:07 AM, james jan wrote: hi all, i've changedallow=all and restarted service. butstill gives488 Not acceptable here The softswitch sends codec g729. "core show translation" says codec g729 alsa installed. On Sun, Jul 28, 2013 at 10:11 PM, Andrew Colin and...@vsave.co.za wrote: 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 it will work but thats not secure at all. How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me. -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailma
Re: [asterisk-users] asterisk ip authentication
No Acceptable here is a codec error. Check the other soft switch and see what codecs it is sending. if you say allow=all it will work but thats not secure at all. Kind Regards 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 -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 it will work but thats not secure at all. How is allow=all insecure? I can see inefficient, but what would make that insecure eludes me. -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Behalf Of Harish Mandowara Sent: Saturday, 11 May 2013 8:15 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi driver not getting install Dear, I have redhat enterprise linux 6.3. snip `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux/driver s/dahdi/firmware' You do not appear to have the sources for the 2.6.32-279.el6.x86_64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux' make: *** [all] Error 2 I'm a debian user after an inplace upgrade of Debian 6.0 to Debian 7.0, but had exactly that last night. From googling I reckon you need to install kernel-headers-2.6.32-279.el6.x86_64.rpm Alec Davis -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 Behalf Of Harish Mandowara Sent: Saturday, 11 May 2013 8:15 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi driver not getting install Dear, I have redhat enterprise linux 6.3. snip `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux/driver s/dahdi/firmware' You do not appear to have the sources for the 2.6.32-279.el6.x86_64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/root/Downloads/dahdi-linux-complete-2.6.2+2.6.2/linux' make: *** [all] Error 2 I'm a debian user after an inplace upgrade of Debian 6.0 to Debian 7.0, but had exactly that last night. From googling I reckon you need to install kernel-headers-2.6.32-279.el6.x86_64.rpm Alec Davis -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] B200p card - use dahdi or mISDN? On 10/16/2012 08:50 AM, Sebastian Arcus wrote: I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one of these - it is possible to use either DAHDI or mISDN with it. Are there any factors to consider when choosing which software to use? Is one better than the other - or does one have features which are not present in the other? I would go for DAHDI so you can use the card like you would use any Digium card. OpenVOX also seems to focus on DAHDI integration. Looking at the OpenVOX site it seems that you will need to use the patched DAHDI from here: http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/ Regards, Patrick -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 State. Hi, I am also facing same issue. Is this resolved? Please reply. Thanks -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 - Non-Commercial Discussion Subject: [asterisk-users] My digium card die? Hi. im starting my first steps with asterisk, but i have a doubt about my card. lspci show the card correctly 01:07.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) but when i run dahdi_scan dont show nothing. How can i detect if really my card die? Im running Asterisk 1.8.12.0 on Ubuntu Server 10.04. Thanks and regards -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Mountifield Sent: 29 June 2012 01:52 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI trunk between Asterisk servers does not work. In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
