similar to: problems with trunks IAX2 and queues

Hi everybody and sorry by the insistence. Nobody has working Tinc Server over a Raspberry in an environment in production? Best regards and sorry again, Ramses De: Ramses II [mailto:ramses.sevilla at gmail.com] Enviado el: martes, 08 de octubre de 2013 17:59 Para: tinc at tinc-vpn.org Asunto: Tinc Server and Raspberry PI (Rev. B). Dear gentlemen, I need configure a VPN

I am curious if anyone has had problems trunking iax2 with 100+ concurrent calls. I am planning on testing this out tomorrow, however I wanted to know if anyone else has had a problem with this prior to my test so I know what to look for if anything is known and what resolutions have been found if there are any known problems. Specifically I am doing this on fbsd 6 with asterisk 1.2.4 using

Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a couple of Asterisk 13.4 servers with an IAX2 trunk between them: [phone1] <--pjsip--> [server1] <--iax2 trunk--> [server2] <--pjsip--> [phone2] With this setup, audio calls work fine, but video doesn't work: I get a black window and if I remember correctly, I was getting white noise in one direction (not sure if the noise thing is asterisk's fault or the

Hi, I am using iax2 trunks between asterisk servers and am having a callerid problem. We are using realtime sip clients distributed between multiple servers. Only in test now but have run into a calleeid problem - the name of the called party is not displayed if the called party is on a different server, it works if the called party is on the same server. On each server sip clients show calleeid

Dear Royce; Did ur problem resolved? Because now I am facing same problem. It look like that it happens with IAX trunk only, but does not happen with IAX endpoints that registering (as trunk does not register, it sends the call directly). My initial analysis that one of the following can help to let the trunks talk: if there is an IAX endpoints registering to the machines, then trunk become

https://bugs.freedesktop.org/show_bug.cgi?id=57278 Priority: medium Bug ID: 57278 Assignee: nouveau at lists.freedesktop.org Summary: [xf86-video-nouveau] flightgear crash when loading scenary Severity: critical Classification: Unclassified OS: Linux (All) Reporter: king.infet at gmail.com

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

I have a network of offices using Asterisk that are connected via IAX2 trunks. The trunks work great for a day or two then for no reason at all one end of the trunk will become UNREACHABLE while the other end is still connected. The only way to fix the problem is to shutdown Asterisk completly then start it backup again. The end that dies is not always the same, some times it is server A and some

I am using Asterisk 1.2.13 with Zaptel 1.2.11, I used to have an old PBX connected to one port and the PRI connected to the other. I'm having serious stability issues with Asterisk on a box that has been rock solid previously. The old PBX died two months ago so one port on the TE210P is now unused but still configured. Also I'm afraid I have upgraded from Asterisk 1.2.9.1 and the old

Hi! I was searching for info about this issue and found almost nothing. So, let's go directly to the matters... - Problem: AutoCAD says "You do not have permission to save to this location." when trying to save the file in the samba share dir. (This problem only occur with AutoCAD.) - Scenary: Running AutoCAD in a WinXP/Win7 PC, opening a DWG AutoCAD file from samba share dir in

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with

I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue with it. We can call 911 which is routed out these new trunks, and it goes to the 911 center, but they are not getting the ANI and hence "no record found". Our LEC is Embarq, and they say they can see the call come in and send: KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST I turned on all the debug

I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current) Today when I setup queues for the first time (with one member in my default queue), I got some really strange behaviour, aside from my hysterical laughing after hearing the default MOH =) I only have one SIP hardphone I'm testing with right now, so I tested using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog

Hey everyone here's my problem. Have a queue configured, it plays the desired recording, checks to see if agents are logged in via agentcallback, forwards the call according to distribution method, times out according to timeout settings, logs out the agent that did not answer, hunts for next agent, logs the rest of the agents out one by one when they don't answer, and drops call into

Scenary: Fedora Core 5 + VirtualimPro Ruber RPM''s : ruby-libs-1.8.4-8.fc5 ruby-rdoc-1.8.4-8.fc5 ruby-mode-1.8.4-8.fc5 ruby-libs-1.8.4-8 ruby-debuginfo-1.8.4-8 ruby-ri-1.8.4-8 ruby-1.8.4-8 ruby-rdoc-1.8.4-8 ruby-devel-1.8.4-8 ruby-mode-1.8.4-8 ruby-tcltk-1.8.4-8 ruby-1.8.4-8.fc5 ruby-irb-1.8.4-8.fc5 ruby-ri-1.8.4-8.fc5 ruby-devel-1.8.4-8.fc5 ruby-irb-1.8.4-8 ruby-docs-1.8.4-8

Just by the way, do anybody knows if call pickup of a call to a sip extension from a mgcp phone is supposed to work (even if sip keeps ringing). The scenary is as follows: 3@mgcp02 (ext 136) calls sip/julia (ext 133) and after It starts ringing 2@mgcp02 (ext 135) dials *8. Nothing happens, only 135 gets congestion tone, 133 keeps ringing and in the asterisk console I get: --

Hi This bug is really crazy, please help me In the follow scenary ATA-186 -> SIP -> Asterisk -> SIP -> ATA 186 : No DTMF gets through * in outbound mode, Sip conf [204] type=friend username=204 secret=somesecretpassword host=dynamic canreinvite=no ; The follow line don't work dtmfmode=rfc2833 nat=1

Hello, I have this setup: Telco --PRI(g1,ext-incoming)--> Asterisk TE405P --PRI(g2,int-incoming)--> Alcatel OXO extensions.conf: [ext-incoming] exten => _X.,1,Noop exten => _X.,n,SetVar(CALLFILENAME=${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP:6:2}/${TIMESTAMP}-${UNIQUEID}-${EXTEN}) exten => _X.,n,Monitor(wav,${CALLFILENAME},b) exten => _X.,n,Dial(Zap/g2/${EXTEN}) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! In the network of my house I was testing the security with my Asterisk installation. The first test that I'm doing is an man in the middle attack. In this scenary, the attacker is a virtual machine that it tries to see the SIP traffic between a PC with a softphone and a Grandstream BT200 telephone. But it draws attention to me between