similar to: Optimize peers registration under jitter/delay.

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

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 ;-----------------------------------------

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

In what way is this a bug in R? It looks like a bug in the package, and as Defn.h is not part of R's API any packge using it is `at risk' (and cannot be installed in a binary-only installation, or even an installed version of R). In particular, Defn,.h depends on config.h, and it seems you installed a binary version of R and used separate sources. I would suggest building R from

Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register => 808:password at as2.xxxxx.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99

Hello! Could you help me with Asterisk 11.21.2 and AsteriskNow platform. The problem is: My Asterisk PBX has SIP (chan_sip) trunk to provider. Asterisk periodically loses trunk registratrion: *sip show registry:* /Host??????????????????????????????????? dnsmgr Username?????? Refresh State??????????????? Reg.Time???????????????? // //X.X.X.X:5060??????????????????? N????? <LOGIN>

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi All I am trying to dial out using SIP and Vonage using the instructions : <a href="http&#58;&#47;&#47;www.voip-info.org&#47;wiki&#47;view&#47;Asterisk&#43;and&#43;Vonage" target="_blank"

Hello all, I'm having a quite simple configuration like: SIP provider <=> asterisk SIP <=> lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net

Quick-n-easy questions: Let's say user raub is added to group nosy using smbldap-groupmod smbldap-groupmod -m raub nosy Now, according to ol' ldapsearch, ldapsearch -vvv -H "ldaps://ldap.example.com" -D "uid=admin,ou=People,dc=example,dc=com" -W -b "dc=example,dc=com" -s sub "(cn-nosy)" group nosy has a dn attribute that looks like this dn:

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

To: asterisk-users at lists.digium.com From: webaccounts173 at jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf

I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56]

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

Hello is it possible to determine the SIP.conf parameters 'defaultexpirty' and 'maxexpiry' on a peer basis ? My default value is 300 seconds, but some specific SIP-clients can only send a SIP REGISTER every 3600 seconds. In current configuration these SIP peers now become "Unreachable" after 300 seconds. Or is there another way to differentiate ? Kind regards.

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last