similar to: Could Asterisk be crashing under high context switches?

Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in "batphone mode", which is "immediate=on" in the DAHDI config. They are set up this way because we are an outgoing call

Hi, Is it possible to create a clear zaptel channel which doesn't require to be picked up? The requirement of my client is to open a clear channel to a recorder which starts recording certain message. Currently the channel which is created by zaptel requires the other end to answer the call, and the recorded can't answer, so the channel get hung up after a certain number of rings. Zaptel

Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service,

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

Hello, I may have found a serious issue with 1.6.1.6. I just compiled it yesterday on our server. When anyone tries to access the name directory through the Directory app, the asterisk process completely dies. Our extensions are in a realtime MySQL table, and the directory has worked fine with previous versions of asterisk. Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct in the CDR. Here is the

Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

Hello, I have a strange problem with one of my Zap channels. A user told me that he was in a voicemail system during a call, hit the Flash button, and the call hung up. Now we get no dialtone on the phone hooked up to the channel. Here's the status of the channel: jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7" Parsing /etc/asterisk/extconfig.conf Channel: 7 File

I'm running the samba-client-3.0.20-0.1 SUSE RPM. I was using the version that came with 9.3 but upgraded to see if this specific problem would go away. Guest access does not appear to be working correctly, and it looks like the problem is due to guest not getting mapped into the LDAP query correctly. Specifically, I can login with local account, join workstation to the domain, browse

Had a fairly horrible lightning storm night before last, and four of eight ports in a 1.4.20 machine stopped answering. In the CLI: budsw*CLI> zap show channels Chan Extension Context Language MOH Interpret pseudo default en default 1 from-zaptel en default 2 from-zaptel en default

hello guys... well as my first post i want to say when i try to install rhino it says: ...The installer requires Windows installer 3.0 or newer... so when i try to download that (as a last resort [Wink] ) it fails... so if anyone can help thats great but if not... well this thread is mainly an error report... Thanks! Trey.

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

Hello, We've purchased a 2nd Wildcard TDM2400P for our asterisk server. I've looked around but I can't find documentation on how to properly setup 2 or more telephony boards in asterisk. In particular, I want to ensure that the boards are assigned the same device channels on boot (for example, if you have two NICs in a Linux box and you want to make sure one card is always eth0

Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road,

Hi, First of all I'd like to congratulate everyone engaged in this great project. I've been following this list since late 2014, when I acquired an APC UPS model BZ1200BR. APC bought a Brazilian company named Microsol, which NUT package had drivers for Solis and Rhino models. Since that time, I started developing a new driver for the models below, as a way to support my own device, but

Hey Aaron, I''m just thinking out loud here, but have you considered the possibility of using the Rhino [1] library to implement javascript support in mechanize? It''d create a jruby dependency for that feature, but still. Just thought I''d bring it up while I was thinking about it. -Mat [1] http://www.mozilla.org/rhino/

I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows...... < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000 EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 025 P/F: 1 < 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since

Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a "graceful restart"). Besides, would a "restart gracefully" actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again