similar to: Latency woes, qos the fix?

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing

I am able to return users and groups using wbinfo -g and -u. Samaba will even allow users to connect that are in our domain. The problem exist while trying to narrow down permissions to a share. [public] comment = Public Stuff path = /home/ public = yes read only = no valid users = @"UFAD\_IFAS-FRE-USERS_autoGS" This does not work. It prompts the end user for a

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

On Saturday, April 18, 2020 5:42:11 PM CEST Joshua C. Colp wrote: > On Sat, Apr 18, 2020 at 8:47 AM hw <hw at gc-24.de> wrote: > > Hi, > > > > how do I make a bug report? I filled in the form to make a report and > > https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues > > reported by me. > > If successful then JIRA will redirect

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Hi, how do I make a bug report? I filled in the form to make a report and https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues reported by me. If someone knows how to get asterisk to re-register when using pjsip after the registration shows as Rejected, like after the internet connection to the VOIP provider goes away (and comes back), please let me know. This bug makes

Hi, examining the network traffic with wireshark shows that asterisk does not set any QoS values at all. What do I need to do to make asterisk set QoS values (on Centos 7)? The wiki says to use vconfig to set QoS values[1]. What does the skb-priority need to be set to? How do you use vconfig on interfaces that are not VLAN interfaces? Is it generally impossible to set QoS values on

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Is it possible to mangle the 802.1p priority bit on a packet as it gets bridged? I can''t find anything in either the iptables or ebtables docs to tell me how it''s done. Regards, Leigh Leigh Sharpe Network Systems Engineer Pacific Wireless Ph +61 3 9584 8966 Mob 0408 009 502 email lsharpe@pacificwireless.com.au web www.pacificwireless.com.au

Hi, I'm trying to configure some Quality Of Service among an Asterisk server with RedHat3 and some IP phones on my LAN. I read about 802.1p (level 2) QoS, using 3 bits of VLAN tag. Two questions: - do I need to use tagged links (trunks) end-to-end? In other words, do all ports on all switches from phones to server need to be configured as 'tagged'? - how can I configure ethernet card

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)