similar to: Spontaneous Hangup occuring

Is it possible to allow distinctive rings work for FXS ports as well? I need a certain FXS extension to ring a distinctive double ring. I modified zapata.conf appropriately for dring1,dring2 and it just Seems to ignore my updates. Do distinctive rings only work for FXO ports? Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

Hello all, I am trying to setup a local asterisk server with SIP & ZAP extentions with And IAX2 'switch' to another Asterisk gateway with a PRI. I have managed to Get it working correctly for calls coming in and out. However, CallerID only Seems to work when I dial out. However, everytime I receive a call, it shows Up as 'IAX Guest User' on a SIP client. How can I

Hello all, I have a WildCard TE410P setup and working with a full PRI with the latest CVS. SIP and IAX2 gateways are accessing the PRI without issue, however; echo is very prominent in some calls and is only heard by the IAX2/SIP client. The echo Is not present in calls to cell phones because they are digital, centrex land Lines have a barely noticable echo, but analog lines aint so pretty.

Hello all, I have a TE410P hooked up to a single PRI. Incomming CallerID is fine, and Outgoing works as well. However, if I change my dialplan for an extension To do a 'follow you, follow me' or setup an auto attendant that rings extensions Thru to cell phones, the CallerID always shows up on the call reciever as '708'. When I do a verbose dump at the console, it appears that

Which CVS branch is the correct official Asterisk -STABLE? cvs up -r v1-0-1 asterisk zaptel libpri asterisk-sounds asterisk-addons OR cvs up -r v1-0 asterisk zaptel libpri asterisk-sounds asterisk-addons -------------- next part -------------- A non-text attachment was scrubbed... Name: eric.vcf Type: text/x-vcard Size: 146 bytes Desc: not available Url :

Hello, Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds. There is now a core dump present in /tmp : -rw------- 1 root root 88M Aug 26 08:07 core.sip.pbx.tld-2011-08-26T08:07:35+0200 How can I get usefull information about what went wrong ? Because a spontaneous reboot of Asterisk has never

Hi Bill, On 2016-05-30, Bill Gee <bgee at campercaver.net> wrote: > > By luck I saw the beginning of a reboot on the server console. Normally I have > other systems up on the KVM switch. It appears to have dumped core. I don't > know where to look for the core dump files. They are not in /root. One place you might check is under /var/lib. I think there may be a

I just want to get this information into the mailing list archives. In my situation I had Windows XP SP2 workstations mapped to a samba share on a debian etch stable server. My corporate overloards changed our AV product to Norton. After that, when users would rename a file or a folder on the samba share the workstaion would suddenly reboot (and not execute the rename). The samba server was a

I'm using the samba 2.0.6 rpm's from rawhide (2.0.6-9) - I modified the .spec to remove the --with-mmap and added syslog support (and put smbspool into the system) I get these errors in my logs: smbd[3713]: getpeername failed. Error was Bad file descriptor smbd[3713]: getpeername failed. Error was Bad file descriptor smbd[3713]: getpeername failed. Error was Bad file descriptor

Hello. A while back, I noticed an odd problem with our SPA-841 phones connected to Asterisk. Now we are having a different odd problem, and I'm not sure if they're related. I wonder if anyone else has experienced anything else like this, and/or if there is any reasonable explanation? Occasionally, one of our SPA-841's will spontaneously start up with "Welcome to Comedian

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri 1.4.7 and I am getting the error: -- Requested transfer capability: 0x00 - SPEECH -- Called 23/317506XXXX -- Channel 0/23, span 1 got hangup, cause 99 -- Hungup 'DAHDI/23-1' on DIALING out. Calling in seems to work just fine. Seems like everything is configured fine... system.conf

Hello everyone - My CentOS 6.8 server has been rebooting itself every 2 to 4 hours for the last several days. I do not know where to look for logs that might give a clue what the problem is. There are no unusual entries in /var/log/messages. I looked over other log files in /var/log and found nothing suggestive. Where else can I look? By luck I saw the beginning of a reboot on the server

hi, Just wanted to inform everyone, if you're using the latest bristuff's you might (depends on the country!) have hangup issues. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Then the remote party keeps ringing. In some situations (we only encountered this while dialing to other * servers) it keeps the line open on the

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

Hi, Although zonedata.c contains ITU E.180 recommendations for Turkey, we are still experiencing unrecognized hangups from Turk Telekom PSTN lines when callers hangup. Turk Telekom does *not* provide supervised disconnects on analog PSTN, and the tone we receive we when caller hangs up is similar to busy, with three short beeps, followed by one long beep, which keeps repeating. We've

I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the