similar to: asterisk boxes looses registration

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

So I have two CentOS 7 machines running. if I am on my server and I do "curl http://192.168.1.8" I get data. If I do "host devgeis.LayeredSolutionsInc.com" I get the correct address 192.168.1.8 if I goto another machine with CentOS 7. I do "curl http://192.168.1.8" I get data. I do "host devgeis.LayeredSolutionsInc.com" I get the correct address

Did you notice the address ? It is not the same IP. Patrick Le 02/11/2020 ? 17:48, Jerry Geis a ?crit?: > So I have two CentOS 7 machines running. > > if I am on my server and I do "curl http://192.168.1.8" I get data. > If I do "host devgeis.LayeredSolutionsInc.com" I get the correct address > 192.168.1.8 > > if I goto another machine with CentOS 7.

I have a cloud server... I have a phone in Chicago I have a phone in Indiana. Both are registered to the cloud server - using chan_sip and Asterisk 18.18.0 I can send a pre-recorded message to Chicago it auto answers and hear audio. I can do the same to the phone in indiana. however - when i call from Indiana to Chicago - the phone rings - but I do not get any audio? I have in sip.conf

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

I have a problem with calls not hanging up if for some reason the physical phone dies or gets unplugged I can demonstrate this in practice by making a call from a handset, then unplugging the handset from the power. The call remains active and asterisk never seems to disconnect it. More annoyingly when power is re-applied the handset comes back to life, won't receive incoming calls

I have had a case where after a hangup on the Alsa channel asterisk still thinks the line or call is active. I have: rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 in my sip.conf file to help with this but it had no effect. How can I ensure a session HANGS up and is not stale???? Is there a way for the next incoming call to VERIFY that console/ALSA channel is still valid. I dont want to

Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

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>

I am trying to boot a grub entry for CentOS 8 menuentry "Server Install CentOS 8" { linux /boot/vmlinuz noverifyssl ks= https://devgeis.LayeredSolutionsInc.com:443/kickstart/ks_update_to_server8.cfg biosdevname=0 net.ifnames=0 ksdevice=eth0 ip=192.168.1.13 gateway=192.168.1.1 netmask=255.255.255. 0 nameserver=192.168.1.1 inst.sshd sshd=1 initrd /boot/initrd.img

Hi, I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the

I am getting these errors on my machine for an external USB connection. Apr 8 13:23:55 devgeis kernel: xhci_hcd 0000:b3:00.0: ERROR Transfer event for unknown stream ring slot 4 ep 3 Apr 8 13:23:55 devgeis kernel: xhci_hcd 0000:b3:00.0: @00000020155b24c0 00000000 00000000 1a001000 04048001 Apr 8 13:23:55 devgeis kernel: sd 7:0:0:0: [sdd] tag#3 uas_eh_abort_handler 0 uas-tag 4 inflight: CMD OUT

Firstly, I'm running Asterisk 1.4.4 on Solaris 10. I have several different internal SIP phones all sharing a single IAX2 VoIP channel. PHONES |------------- <SIP/uLAW> --------------| ASTERISK |-------------- <IAX2/g729> ------------|VoIP/ISP The g729 codec has been registered successfully and appears to be detected by Asterisk (NOTE: I have changed what I thought might have

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem

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 ~]#