similar to: POlycom phone not ringing behind firewall (401 permission denied)

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi Steve, - Your right - the file was AMI (copied the other one). By direct connect I simply meant the speaker is an extension on that server. here is the SIP debug <--- SIP read from UDP:X.X.X.X:1024 ---> == Using SIP RTP CoS mark 5 Audio is at 16060 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably

Endpoint sends an INVITE Asterisk send an INVITE to the Carrier Carrier is down, does not even sends ACK PJSIP sends several INVITES End point sends <--- Received SIP request (397 bytes) from UDP XXXX::50187 ---> CANCEL sip:xxxxxxx at xxxxxxx SIP/2.0 Via: SIP/2.0/UDP xxxxxxx :50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport Max-Forwards: 70 To: <sip:xxxxx at xxxxx> From:

Hello Dear Friends, it is CentOS Release 6.2, ntpd is running but do not see bounded to the port udp:123 any guidelines would be very much appreciable. [root at jet mavi]# netstat -ntlp | grep ntpd <nothing in output> [root at jet mavi]# netstat -ntl | grep 123 <nothing in output> But the service is running - [root at jet mavi]# /etc/init.d/ntpd status ntpd (pid 15876) is

Hi all, I'm a fairly long time user of Asterisk, but I'm new to this list. I used to use the old forums some few years ago. I wanted to ask why there are different Asterisk versions, as shown by the announcements in the past week or 2: Asterisk 13.33.0 Asterisk 16.10.0 Asterisk 17.4.0 I'm currently using 16.8.0 and wondering if I should upgrade to 16.10.0, or perhaps give 17.4.0 a

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 have a problem, problem is slow samba write performance.I tested samba copy and ftp copy, result and machine spec; Samba PC P4 3 ghz / 2 gb Ram 1 x 80 gb HDD(for system) 4 x 250gb sata hdd (software raid /dev/md0)^ 3com gigabit ethernet Windows Pc AMD 2500+ / 5123mn ram 1 x 80 gb hdd (c:\) 1 x 160 gb hdd (d:\) gigabit ethernet test results; windows hdd to hdd copy Source File Size

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

All, Our project has modified the kernel and while testing it with xen, specifically running virt-install to install a domain, the command hangs when the install kernel is booting: # virt-install What is the name of your virtual machine? dom1 How much RAM should be allocated (in megabytes)? 500 What would you like to use as the disk (path)? /export/home/dom1 How large would you like the disk

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all &Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten =>

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate

Hey Dave, In the case of 13 and 16, these are LTS versions which means that they get long term service. 17 is a standard release. The benefit of an LTS is that you can expect it to get bug fixes and improvements for an extended period of time without anything major being changed. If you find an LTS version that has everything you need, it's probably the safest version to choose. Any

Thanks for that info, Ben. I do like to test out the latest and most up-to-date versions of things when I can, so I'll check those files and see how it goes. On 2020-05-11 17:20, Ben Ford <bford at digium.com> put forth the proposition: > Hey Dave, > > In the case of 13 and 16, these are LTS versions which means that they get > long term service. 17 is a standard release.

I am aware that there have been several posts related to ssh connections hanging, i have tried to read through most of them in the archives... I am posting this in hopes that it may present something new that will further help resolve this problem. The problem i am having appears to be similar to what others have reported where after typing exit in an interactive shell the connection is not

Is there any way to have more specific disposition codes in the CDR? Currently there are only 3 values: ANSWER, NO ANSWER, BUSY. In this way, when i call a cell phone that is switched off i get "NO ANSWER", while i would like to be able to log that the call is not answered because "The customer you have dialed is unavailable at the moment". The same for "non

Installed uploadify_rails3 gem, while submiting the form all params are correct but I am getting error "Completed 401 Unauthorized in 0ms" which prevents me from doing any further steps. If any of you has faced such situation please send me any solution to get rid of it. Thanks in advance -- You received this message because you are subscribed to the Google Groups "Ruby on

Hi, I''m having some troubles related with reports. Catalogs are updated between server and clients but I cannot submit reports because of the 401 error. All is caused by the security that I had to put and I don''t know how to solve it. Anybody knows how to fix it? Thanks in advance. -- You received this message because you are subscribed to the Google Groups "Puppet

Hi all, I've upgraded from 2.0.x to 2.2.2 on a Tru64 5.1 box. I try to start SWAT with root and roots password and get a : 401 Bad Authorization username or password incorrect With a non root account I can access, but not manage the server. What' wrong? Thanks in advance Roman