Re: [asterisk-users] Paging systems?

to the desired zone. the page would complete when the call is hung up. You would likely need to make sure the ATA is using current loop disconnect or reverse to ensure hang-up. I think it should be the PABX config using the Figure 3 configuration. Best of luck Bryant Zimmerman Sr. Systems

Re: [asterisk-users] Paging systems?

handle being the amp for a few speakers. Bryant Zimmerman Sr. Systems Architect Grand Dial Communications, A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) From: Darryl Moore Sent: 3/21/19 4:59 PM To: Asterisk Users Mailing

[asterisk-users] Asterisk 13.18.4 - New Error PJLIB_UTIL_EDNS_REFUSED

stem seems to be working find. Anyone have an idea what could be triggering this issue? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] user-agent access from pjsip

I am trying to get the user-agent from extensions registered via pjsip. With sip we could do a sip show peer peername and it would list the user-agent string. In a pjsip deployment it looks like this info is likely in the contact. I know we can access it from the dialplan, but this is only

[asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only

?We have upgraded a system from Asterisk 11 to Asterisk 13 with pjsip. We are experiencing random Jitter on outbound calls. This was not occurring when running asterisk 11. We have two IP's bound to pjsip one on the private vlan network the phones are on and the asterisk one on the asterisk

Re: [asterisk-users] PJSIP add header not working

}"="1"]?addSessionCallInfo,1) exten => ThisHeader,1,Set(PJSIP_HEADER(add,ThisHeader)=ValueToSet) exten => ThisHeader,n,Return() exten => ThatHeader,1,Set(PJSIP_HEADER(add,ThatHeader)=ValuetoSet) exten => ThatHeader,n,Return() exten => addSessionCallInfo,1,Se

[asterisk-users] Bug in func_odbc module

Hey all I have code we are moving from an early asterisk 13 system to the latest build. The issue we are having is func_odbc calls are acting incorrectly. We have tables that have fields with null values in them. On the new system when we read a field with a null value it is

Re: [asterisk-users] Asterisk pjsip registration issues - Solved

Dave from_user fixed the issue. Thank You Thank You Thank You I was about ready to chuck pjsip. The lack of good / complete documentation is a real problem. Man you saved me another late night. Thanks Bryant From: "Dave Platt"

[asterisk-users] Asterisk pjsip registration issues

Hey all I am hoping someone can assist I have now spent over a week trying to figure out what is going on with PJSIP registrations. I am able to register handsets against an asterisk 13 server running pjsip, but I am not able to get pjsip to register out to an older chan_sip asterisk

Re: [asterisk-users] Registering Asterisk 13 server PJSIP to Asterisk 11 SIP

Hey all I am trying to register a PJSIP server on our office to an Asterisk 11 chan_sip server in a datacenter. I keep getting WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': Unable to create

Re: [asterisk-users] Realtime pjsip issues

Original Message > From: "Joshua Colp" <jc...@digium.com> > Sent: Friday, September 15, 2017 11:31 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Realtime pjsip issues > > On Fri, Sep 15, 2017, at 12:18 PM,

Re: [asterisk-users] Realtime pjsip issues

onfig.config sorcery.conf Thanks Bryant From: "Joshua Colp" <jc...@digium.com> Sent: Friday, September 15, 2017 9:56 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime pjsip issues On Fri, Sep 15, 2017, at 10:3

Re: [asterisk-users] Realtime pjsip issues

. Thanks Bryant From: "Joshua Colp" <jc...@digium.com> Sent: Thursday, September 14, 2017 4:34 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime pjsip issues On Thu, Sep 14, 2017, at 05:27 PM, Bryant

Re: [asterisk-users] Realtime pjsip issues

[asterisk-users] Realtime pjsip issues On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote: > This appears to be some kind of cache issue. > We have been doing caching with earlier versions of asterisk 13 on the > pjsip realtime, but now for some reason > The items only show up the

Re: [asterisk-users] Realtime pjsip issues

suggestions. What are others really seeing? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Bryant Zimmerman" <brya...@zktech.com> Sent: Thursday, September 14, 2017 2:43 PM To: asterisk-users@lists.digium.com Su

[asterisk-users] Realtime pjsip issues

We are having an issue where on the latest version of asterisk when configuration pjsip via realtime. we do a pjsip list endpoints it shows our endpoints but lists them as invalid. When we do the pjsip list endpoints again it shows no objects. This applies to pjsip list aors as

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

Thomas Bria is by counterpath Bryant From: "Matt Riddell (lists)" Sent: Saturday, April 29, 2017 11:50 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re:

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

out you are looking for quality on the cheep. Desk phones are cheep and in most cases just work and offer consistent quality. If others have found different I look forward to seeing their responses. This is a great question thanks for asking it Thomas. Best of luck Bryant Zimmerman (ZK

Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]

In most instances the company being called is not charging the caller for their phone serves. That is the callers service provider, and once the answer is issued the call is up. This only makes senses if the company being called is providing services and charging a per min rate for that

Re: [asterisk-users] WebRTC - Transport Issues. - Solved

gt; Sent: Sunday, March 12, 2017 7:35 PM On Sat, Mar 11, 2017, at 09:52 PM, Bryant Zimmerman wrote: > Hey all. I have webrtc up and running with asterisk 11. All is going well > with TLS now working. > At least I hope it is using TLS and wss. Based on what I am seeing I > have >

[asterisk-users] WebRTC - Transport Issues.

Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I

Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code. - Solved

I figured this out. I had to set the outofcall_message_context = messages on the actual peer. It was not good enough to set in the sip.conf Thanks Bryant From: "Bryant Zimmerman" <brya...@zktech.com> Sent: Friday, March 10, 2017

Re: [asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

ons.conf, define a context "messages" with the appropriate extensions (to stick to your example, it will be 16162995607) and use the function MESSAGE to retrieve the SMS content. Best regards Jean Aunis Le 10/03/2017 à 00:21, Bryant Zimmerman a écrit : I am trying to send SMS from my gra

[asterisk-users] Trying to get SMS from GXV3240 to trigger dialplan code.

I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a

Re: [asterisk-users] fail2ban Asterisk 13.13.1

John V Are you using pjsip? We are have several test servers and I just checked my /etc/fail2ban/filter.d/asterisk.conf and it is not updated for pjsip implementations. Looking at the security log files and the regex I noticed that some items are being banned but others are not due to

Re: [asterisk-users] pjsip realtime - endpoints not loading - Solved

It appears that res_odbc.so does not always load fast enough to allow the realtime mappings in the extconfig.conf to complete successfully at startup thus stopping the first load of the pjsip endpoints and other pjsip values. The resolution for this is to preload the res_odbc.so and

[asterisk-users] pjsip realtime - endpoints not loading.

We are continuing to test our asterisk 13 pjsip deployments. I am running into an issue that I am assuming is a configuration problem, and am hoping someone can point me in the right direction. We are running pjsip in real-time mode using a database to store all the endpoint records. Our

[asterisk-users] Fax faling on PJSip

I am working on moving from version 11 to version 13 for my fax applications. We are bumping into an issue where the bulk of the T38 faxes are failing. The sending test switch is reporting COMREC_ERR_TRANSMIT_PHASE These same faxes succeed on the 11 version of asterisk. I am wondering if

Re: [asterisk-users] Asterisk 13 T.38 Version 3?

Does anyone know if Asterisk 13 will support T.38 Version 3? ? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] pjsip transports from database.

On Friday, November 4, 2016 10:20 AM - Joshua Colp wrote: >>On Fri, Nov 4, 2016, at 10:26 AM, Bryant Zimmerman wrote: >> Hey all >> >> I am trying to configure all my pjsip transports form a database table. >> The issue I am running into is that pjsip is

[asterisk-users] pjsip transports from database.

Hey all I am trying to configure all my pjsip transports form a database table. The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 before it reads my list of transports from the database. This means that my entries for port 5060 are already bound and the settings in

Re: [asterisk-users] PJSIP - State of the art

I agree the multi-domain environment is a nice idea, but too many endpoints don't properly support. We to use a prefix in the SIP username for multi-domain environments. Thanks Bryant From: "Ludovic Gasc" Sent: Sunday, July 17,

Re: [asterisk-users] Call File - CPU spikes

I am working on a project that we are seeing a 100% CPU spike when we move 50 calls files to the folder. We are running pjsip and asterisk 13..It holds the spike for several minutes Are there any tunable that may help with this? Thanks Bryant --

Re: [asterisk-users] ? Re: Recommendations for free virtual server tech and Asterisk? (Ikka Tirtawidjaja)

Hyper-V works well we run both OpenSuse and Debian with asterisk on it is rock solid, and it is free if you use the Hyper-V Server Version. Bryant From: "Saint Michael" Sent: Saturday, April 9, 2016 1:23 PM To: "Asterisk Users

[asterisk-users] Asterisk 13 - Call Bridge issue.

Even when using the U option just issuing the Answer does not seem to always work. I end up having to play a prompt of some sort to force the answer.. There has to be some kind of bug going on here. Thanks Bryant From: "Bryant Zimmerman&q

Re: [asterisk-users] Asterisk 13 - Call Bridge issue.

From: "Bryant Zimmerman" <brya...@zktech.com> Sent: Thursday, March 31, 2016 6:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 13 - Call Bridge issue. I have the following scenario. Call file

[asterisk-users] Asterisk 13 - Call Bridge issue.

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing

Re: [asterisk-users] 2 devices same *actual* extension - can it be done

With Asterisk 13 you may be able to do it with PJSIP using two separate connections on the same AOR I believe you would have two separate endpoints that would register under the same user and auth. If I understand it correctly when you send a call to the AOR both registered endpoints would be

Re: [asterisk-users] Grandstream Early Dial

try this as we are a heavy grandstream shop. It has been something on the list. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Jean-Denis Girard" <jd.gir...@sysnux.pf> Sent: Friday, February 19, 2016 11:53 AM

Re: [asterisk-users] Grandstream Early Dial

Jean-Denis Girard I have not used the Incomplete yet, but you might be able to do something like this. [earlydial] exten => _.,1,Set(l_Extension = ${EXTEN}) exten => _.,n,Goto(${l_Extension},1) exten => _.,n,Goto(noMatch,1) exten => i,1,Goto(noMatch,1) exten => noMatch,1,

Re: [asterisk-users] How to execute a macro after dial but before connect

Phillip Check out the b and B options one of them should do what you want. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Saint Michael&q

Re: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones

Richard Check both the DTMF settings, and the DialPlan string for account 3 on the phone. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Richard Schroeder" <rsch...@gmail.com> Sent: Tuesday, February

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

Sonny We use a real-time database for adding pjsip users. If you want to do it from the pjsip.conf you would have to write to the file from a script of some sort and then trigger a reload. There is a real-time implementation for the extensions.conf as well. I personally use scripts for

Re: [asterisk-users] PJSIP Stun/ICE

George Reloading transports is one critical part and it sounds like you are making headway on that. I have yet to be able to get transports to load from a real-time table using sorcery.conf If I would get the transports pulling from real-time as the (documentation says is possible but I

Re: [asterisk-users] PJSIP Stun/ICE

the board? Thanks Bryant From: "Joshua Colp" <jc...@digium.com> Sent: Tuesday, January 26, 2016 8:10 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk

[asterisk-users] PJSIP - Realtime - Transports module?

Does anyone know which module the type=transport loads under. I am trying to set up transports to load from a realtime table. I added the following under [res_pjsip] and it does not poll the associated database. [res_pjsip] transport=realtime,vap002_ps_transports We also set the

Re: [asterisk-users] PJSIP Stun/ICE

hrieb Joshua Colp <jc...@digium.com>: > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is pulled from the transports table in realtime >> during asterisk startup it can't get any updates? >> Can a new transport be added to the table and the associated endpoi

Re: [asterisk-users] PJSIP Stun/ICE

Joshua I look forward to improvements as time goes on with PJSIP. I have been trying all day to get the Transport objects to pull from a real-time table. The documentation says it is possible, but does not show any examples. I am hoping to have the Transports pulled from the table at

Re: [asterisk-users] PJSIP Stun/ICE

ion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] PJSIP Stun/ICE Bryant Zimmerman wrote: > I have an asterisk 13 server behind NAT on a dynamic IP Address. It is > running the PJSIP Stack > It is registering to another asterisk 13 server that is on a Static IP

[asterisk-users] PJSIP Stun/ICE

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind

[asterisk-users] PJSIP NAT traversal.

I have two servers running pjsip they are both on NAT. The proxy has a static public address. I set the ;external_media_address=203.0.113.1 and ;external_signaling_address=203.0.113.1 to the actual IP address in the transport section on the proxy. The issue I am having is on the server

Re: [asterisk-users] Using external RTP proxy for res_pjsip

Dmitrity What kind of volume are you running? You can use asterisk as a proxy if you set it up correctly. The choice would fall on the volume and the operational needs. To use an external proxy you would either need to register to the proxy or have a trusted IP to IP relationship. If your

Re: [asterisk-users] Remote UNIX connection / disconnected.

Anyone know how to suppress the -- Remote UNIX connection / disconnected messages. I have a monitoring application that calls asterisk from the command line to verify some uptime stats. I would like to not have the console log the connections.. Any ideas are appreciated. Thanks Bryant --

Re: [asterisk-users] pjsip show xxxx like endpoint?

George, and Matthew I can open an issue later today, but if you want to do it that would be awesome as well. Please post the issue number back to this thread so I can follow it. Ideally the Like would work with all pjsip show commands so we can reduce the list and drill down just like

Re: [asterisk-users] pjsip show xxxx like endpoint?

George and Mat Here is the link to the Jar Issue. https://issues.asterisk.org/jira/browse/ASTERISK-25477 Thanks Bryant From: "George Joseph" Sent: Sunday, October 18, 2015 10:17 PM To: "Asterisk Users Mailing

Re: [asterisk-users] pjsip database error when using MS SQL via ODBC

unsure of the long term issues associated with this. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] pjsip show xxxx like endpoint?

Is there a way to limit the items returned by pjsip show [type] using like chan_sip allowed for sip show peers like , but I can't seem to figure out how to lookup or limit my returns with pjsip Thanks Bryant -- _ --

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

From: "Ryan, Travis" Sent: Monday, October 5, 2015 8:20 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] pjsip realtime registrations not pulling

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

From: "Joshua Colp" <jc...@digium.com> Sent: Monday, October 5, 2015 9:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-05 10:15 AM, Bryant Zimmerman wrote: > > -- > I am working o

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

I have a pjsip install that is not pulling it's realtime registrations from an ODBC database. When I have them directly pull from a MySQL database everything seems to work. I am having trouble finding a requirements document for the pjsip realtime registrations. Is there some kind of

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

On 15-10-04 09:54 AM, Bryant Zimmerman wrote: > I have a pjsip install that is not pulling it's realtime registrations > from an ODBC database. > When I have them directly pull from a MySQL database everything seems to > work. > I am having trouble finding a requirements documen

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

From: "Joshua Colp" <jc...@digium.com> Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

From: "Joshua Colp" <jc...@digium.com> Sent: Sunday, October 4, 2015 12:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:42 PM, Bryant

[asterisk-users] Asterisk Qualify to pjsip

I am running a pjsip test between two servers one running pjsip and one running chan_sip The chan_sip side is sending requests based on qualify=yes. The pjsip side is showing notices.. Exp ?[Oct 4 18:09:02] NOTICE[5982]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request

Re: [asterisk-users] Asterisk AMI events filtering

Sam Based on my experience you need to write a middle tier that has what you want exposed to the users.. AMI was not really designed to offer direct multi-tenant access. That is for your middle tier to handle. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003

Re: [asterisk-users] Asterisk SMS

://www.voip-info.org/wiki/view/Asterisk+cmd+Sms Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Distributed Device States - Best Option

We have used AIS for disturbed Device State in the past, BLF and MWI, We are in the process of an update on one of our clustered systems, We are looking at XMPP and I found a few discussions on a Corosync with has OpenAIS built in. My question is which should I be looking at to replace my

Re: [asterisk-users] adding area code

Thanks for your reply, [globals] AREACODE=381 [outbound] exten = _9XX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN-1},80) did not work for me, any ideas? Thanks, On 04/27/2015 01:59 PM, Phil Reynolds wrote: On 27 April 2015 21:32:42 BST, Motty Cruz motty.c...@gmail.com wrote: Hello,

Re: [asterisk-users] adding area code

code and the last 7 digits of your dialed phone number exten = _9XXX,n,Dial(SIP/${dialnumber},35) Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Motty Cruz motty.c...@gmail.com Sent: Monday, April 27, 2015 4:33 PM

Re: [asterisk-users] FXO advice

Alejandro All of the Grandstream devices can be remote provisioned if you know what you are doing. Bryant From: Alejandro cdgr...@gmail.com Sent: Wednesday, April 15, 2015 4:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users]

Re: [asterisk-users] Grandstream GXP2140

We use a lot of GXP21x phones. We have had issues with the GXP-2140 when using the side car as BLF's. The device becomes sluggish after about 45 days of operation a reboot solves the issues. This has been reported but not resolved as of yet.. If you are not using the side car the issue does

Re: [asterisk-users] Fidelio protocol and Mitel protocol

Does anyone know anything about the Fidelio and Mitel protocol for hotel / motel? Are these industry standards or proprietary formats? Are there open standards for communication with Hotel management software's that could be used in conjunction with a custom asterisk deployment?

Re: [asterisk-users] switching from SIP to Skype..or not

Hey all We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases

Re: [asterisk-users] Unstable phone connection

AGL helpers on the router. Make sure that the site is not double NATing Try using a stun server and see if that helps at all. Watch you console on your sip serer to see how long the device runs before losing connection. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003

Re: [asterisk-users] GXP 1405 and asterisk

with the info value of ring3 is matched Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: ricky gutierrez xserverli...@gmail.com Sent: Thursday, March 12, 2015 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk

Re: [asterisk-users] func_odbc 123

a restart. Thanks Bryant Zimmerman (ZK Tech Inc.) From: Thufir hawat.thu...@gmail.com Sent: Tuesday, March 10, 2015 4:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] func_odbc 123 with func_odbc, in the definitive asterisk guide

Re: [asterisk-users] New Asterisk build

John I will have to get one of these and give this a try. Thanks for sharing. Thanks Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.) 616-855-1030 Ext. 2003 From: John Novack SCII jnov...@stromberg-carlson.org Sent: Friday, March 6

Re: [asterisk-users] New Asterisk build

options. There are several other low cost asterisk like PBX's out there as well, Allo and several others, but I know the GS options work) Good Luck and I hope this info helps. Thanks Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.) P.S. Glen's post also offers some good

Re: [asterisk-users] Investigating international calls fraud

If you have not done so contact the carrier immediately. Report the fraud. Have them disable international on the account until you have your security issues addressed. Ask them to pull call logs containing Source and destination IP address. for the fraud calls. If you are sure they came from

Re: [asterisk-users] Return SIP 401 on hangup

I am having and issue I hope someone can help with.. I have calls that often come in that need to be blocked. We wish to do this without answering the call. The issue is our carriers have fail over servers and will try sending the call from each when we block the call. If we send a

Re: [asterisk-users] Confbridge

dynamically it does not override what is set on the channel. exten = s,n,Set(CONFBRIDGE(user,music_on_hold_class)=latin) Does anyone have any ideas on how I might fix this as well? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003

Re: [asterisk-users] Astricom 2014 presentations

On 10/29/2014 05:50 AM, Bogdan Cristea wrote: Hi Will the presentations made at Astricom 2014 be made public as recorded videos ? thanks Bogdan I'll second the request for that, and also ask if the sessions on Kamailio will be similarly available. Cheers, j That would be awesome if

Re: [asterisk-users] unidata incom ICW-1000G - On asterisk

I am trying to use an ICW-1000G wireless handset connected to an asterisk server remotely The user is working from an offsite location and it appears that the device is not sending out keep-alives or stun. The manufacture is not being of assistance at all. I am wondering if anyone has

Re: [asterisk-users] AGI scripts - delay issue.

Hey All We have several AGI scripts that access databases. These work well most of the time. The issue we are having is that on rare occasion our script must fail to a backup database server. When this occurs it may take up to two seconds to do so. The issue is when there is this

Re: [asterisk-users] Dynamic Parking Lots. Music on Hold Class

How can we set the music on hold class using the Dynamic Parking lots? The variables set the PARKINGLOT, PARKINGDYNAMIC, PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT I can't find a PARKINGMOH variable. This is becoming a big issue. We are using the current release 11. version We have

Re: [asterisk-users] AGI script VERBOSE cmd

Message to send 4) Any ideas what I might be doing incorrect? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] AGI script VERBOSE cmd

Hey all Please disregard my question. I was looking for the word Verbose to show up. I was just being dense. There was no real issue it is working just different than what I was expecting. Thanks Bryant From: Bryant Zimmerman brya

Re: [asterisk-users] Live Recording on the Storage Server?

for each type of task. In some cases our .delete files are processed as moves to an abandon cache for recovery if a customer did not intend to abandon it. The sky's the limit on how complex you want to make it, but in the long run it is fairly simple and it just works. Thanks Bryant Zimmerman

Re: [asterisk-users] func_odbc

Hi All Anyone know how to do include files with func_odbc.conf? I now have several pages of functions in my func_odbc.conf and it is getting harder to maintain it. I would like to break them up into files by category. The standard method of using the #include does not seem to work .

Re: [asterisk-users] Asterisk SIP server on windows

On Wednesday 04 December 2013, Ruddy Gbaguidi wrote: Hi all, I need to build an application that will be an SIP server program that will run on Linux and Windows. The sip server need only some features such as be able to : - Register sip endpoints - Answer a call

Re: [asterisk-users] Asterisk SIP server on windows

the user and posts it to our databases. The default image then has a script that pulls the info down (images uses DHCP to start) and re-writes the asterisk configs. This process is not a small task but if you have the time and budge it can work very well. Thanks Bryant Zimmerman (ZK Tech

Re: [asterisk-users] Voicemail greeting playback issues?

From: Doug Lytle supp...@drdos.info Sent: Monday, November 25, 2013 6:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail greeting playback issues? Bryant Zimmerman

[asterisk-users] Voicemail greeting playback issues?

goes away. Any Ideas? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Voicemail greeting playback issues?

files to and from the local storage. I forced off g729 to ensure that it was not causing the issues. Do you know of any way to force a higher level of debugging to see why the voicemail application would be having an issue? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003

Re: [asterisk-users] Voicemail greeting playback issues?

From: Bryant Zimmerman brya...@zktech.com Sent: Monday, November 25, 2013 2:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail greeting playback issues? From: Doug Lytle supp

Re: [asterisk-users] 11.6 voicemail message cropped off?

Hey all I am running 11.6 and when a caller is sent to vociemail the greeting is cropped off and the beep occurs quickly. Incoming calls are g729 and this occurs where it is using the default greeting or a user provided greeting. I really want to go production with this are there any ideas

Re: [asterisk-users] 11.6 voicemail message cropped off?

Update When no greeting is recorded the default you have reached ext # greeting is cropped. When there is a greeting it is just ignored and not played at all. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Bryant Zimmerman brya

Re: [asterisk-users] Movistar sip Mexico

Can you funnel them through a specific inbound dial context. Then force a re-invite to g729? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Damian Gonzalez dgonza...@denwaip.com Sent: Thursday, November 21, 2013 8:25 AM

Re: [asterisk-users] is g729 codec free? or under license???

of needing higher bandwidth between the client endpoints and the phone. Figure about double the bandwidth when using this method. It may or may not be worth it to you depending on your scenario. Please let us know if this information helps you. Thanks Bryant Zimmerman Sr. Systems

Re: [asterisk-users] Conf Bridge

Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10

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