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

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] 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] 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] 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] 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] 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

[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] 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

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] 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

[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] 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

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

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