Asterisk GUI in Version 1.6

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The ISO download appears to have Astersik 1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI. The only option for the Asterisk GUI is to use 1.4. Is it as simple as installing 1.6 only then using the yum repository to install the Asterisk GUI? If so, what packages are needed? Thanks!  

Asterisk Users 5.1 years ago 6 Answers

Asterisk and Cisco 3825 with ISDN

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I recently subscribe for an ISDN and terminate it on a Cisco 3825 router. I used it as a sip trunk for my asterisk. i'm a newbie when it comes to ISDN. and i've been experiencing some issues: 1. Call Hangup: When hangup is initiated from the outside the extension (softphone/ip phone/VoIP Phone) does not hangup, is this normal? shouldn't asterisk hangup the extension as well when it receives the hangup properly from ISDN? or maybe it's because asterisk does not detect the proper hangup? 2. Caller-ID for incoming calls: When i call in to the ISDN using my hand phone there are times that…

Asterisk Users 5.1 years ago 2 Answers

Asterisk IAX2: Separate Signaling and Media

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Supposedly, IAX2 now has the ability to operate with signaling and media in separate streams, very much like SIP. I've read about this feature here[1] and there[2], but I have yet to see how to actually implement or test it. There are no options in the iax.conf sample configs with Asterisk. All suggestions welcome, except those telling me to jump off a bridge because separated signaling and media makes IAX pointless when compared to SIP. :-)  

Asterisk Users 5.1 years ago 4 Answers

Asterisk, HylaFax and Cardiff

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I'm looking for a way to use our implementation of HylaFax on Asterisk with Cardiff (an old installation of Cardiff document stuff). Is someone doing that? If no one has direct experience, is there a HylaFax client that emulates WinFax print-to-fax?  

Asterisk Users 5.1 years ago 5 Answers

All Phones Ringing When Temporary Loss Of Internet

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One of our clients is connected to our Asterisk installation, from two sites, via VPN which works great. Every so often one of the sites VPN tunnel goes does for say a couple of seconds. When that happens all the extensions, including both sites, ring which is bizarre. Has anybody seen this before ? I only see two places in the dial plan where all phones are called; and neither should be tripped :(

Asterisk Users 5.1 years ago 2 Answers

Asterisk Server Hangs After Bridging 2 Channels

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If I use either the Bridge() app, or the manager Action: Bridge() in a certain scenario (Basically to bridge 2 SIP channels, like an attended transfer, resulting in 2 other SIP channels being discarded) then the whole server locks solid. The console stops, the network stops, something is hammering the box and nothing (including debug tools) seem to be able to do anything about it. If I 'nice' asterisk to lowest priority, and 'nice' a copy of 'top' to highest priority, everything still locks. After a short period, the box recovers, seemingly due to the 60 second RTP timer. Anything that was being logged is lost. My theory…

Asterisk Tips 5.1 years ago 7 Answers

Make A Transfer For External Line In Asterisk

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We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A (FXS-1) --- call ----> B(FXS-2) 2) B (FXS-2) press #1 (blind transfer) after that press dtmf 9 (to dial FXO-1). 3) B hungup. 4) A ---- connected to ---- linea externa…

Asterisk Users 5.1 years ago 5 Answers

How To Detect Zombies SIP Channels In Asterisk

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From time to time, I have to manually kill some "frozen" calls with soft hangup commands in my Asterisk PBX phone system. (As far as I can tell, those freezes occurred after network breakdown (VPN or ethernet link between 2 LAN switches). So at this point, I would say I can't do much to keep those network breakdown to happen). So, which tools are available to automatically detect that SIP channels are up without but no RTP media is flowing in or from them ? Regards

Asterisk Users 5.1 years ago 2 Answers