similar to: Asterisk 12.1.1 - Having trouble setting up PJSIP

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-Answer: normal); Any other ideas? Leandro PS I have set

Hello list, I have Asterisk running well inside our network. I did some experiments exposing it to internet but had some issues: 1. NAT issues (voice one way, etc). From what I understand double-NAT users will always have something like this 2. Immediately I see people trying to hack into. I did configure Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc So.. I ended up closing

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone.

I place a call into Asterisk (from SIP phone) and the To header does not have a tag. Asterisk then sends it's Trying response, still no tag in the To header. The phone then replies with OK, this time the To header includes a tag. Is there any way to retrieve this response To header (including the tag field) from the dial plan? I have tried the PJSIP-HEADER read of the To header, but it

I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext => 700 parkpos => 701-702 context => parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700 at 10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701 at 10.0.1.103 Key4: fnc=blf+sd+cp;sub=702 at

We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the

If you have experience with these phones... We are trying to figure out how to transfer an established call on the SPA504G while a second call is incoming. At present, the receptionist has to answer every single incoming call before the XFER softkey is seen again. This is completely unpractical for a receptionist that may have 4 or more calls coming in at the same time. When the

exten => 179,1,SIPAddHeader(Call-Info:\;answer-after=0) exten => 179,2,Page(SIP/180&SIP/181&SIP/182&SIP/184) The asterisk11 page() application works great, but I've just learned that the person who initiated the page can transfer or conference the page if they don't hang it up before using those functions. It never would have occurred to me to try it, but a

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no

Hi, Dovecot version : 2.2.22 (fe789d2) Operating system : DISTRIB_ID=Ubuntu DISTRIB_RELEASE=16.04 DISTRIB_CODENAME=xenial DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS" CPU architecture : Linux 4.4.67-1-pve #1 SMP PVE 4.4.67-92 (Fri, 23 Jun 2017 08:22:06 +0200) x86_64 GNU/Linux FIle system : local UID GID Aug 17 11:47:28 azizee dovecot: imap(jra11[*5063*:*5011*]):

Hello, I have a problem with the import of a date file. I seems verry tricky. I have a text file (end of the mail). Every file has a different number of measurments witch start with "START OF HEIGHT DATA" and ende with "END OF HEIGHT DATA". I imported the file in a matrix but the letters before the numbers are my problem (S= ,S=,x=,y=). Because through the letters and the

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hi all: I am using the libvirt version that comes with Ubuntu 18.04.3 LTS. I have written a script that backs up my virtual machines every night. I want to limit the amount of memory that this backup operation consumes, mainly to prevent page cache thrashing. I have described the Linux page cache thrashing issue in detail here:

On 1/22/20 11:11 AM, Michal Privoznik wrote: > On 1/22/20 10:03 AM, R. Diez wrote: >> Hi all: >> >> I am using the libvirt version that comes with Ubuntu 18.04.3 LTS. > > I'm sorry, I don't have Ubuntu installed anywhere to look the version > up. Can you run 'virsh version' to find it out for me please? Nevermind, I've managed to reproduce with