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Hi. I've got a working dial plan on my home system but there are problems with it and I was hoping someone more comfortable with dial plans might be able to help. In a nutshell here's what I'm currently doing on an incoming outside phone call [default] Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 exten => s,1,NoOp(Answering in default context) exten =>

...ps://github.com/asterisk/asterisk/blob/master/configs/samples/pjproject.conf.sample by few examples try to explain  what usefull info i can get set [startup] log_level=6 type=startup and dig what's usefull is not very productive btw we are using tools like sipcapture.org,voipmonitor.org, callstats.io, elasticsearch+filebeat, ... but without informations whats happening inside asterisk is harder to solve problems Dne 12/12/2019 v 16:00 Joshua C. Colp napsal(a): > On Thu, Dec 12, 2019 at 10:57 AM marek <cervajs64 at gmail.com > <mailto:cervajs64 at gmail.com>> wrote: >...

Hello, I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: The setup: A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15. Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly. Grandstream gxp-2000 as a SIP phone, and a normal mobile

Hi List! I'm trying to get the Kirk IP600 (DECT Wireless phones) to work with * using the Skinny protocol (chan_sccp doesn't work, the phones do not register and I don't know how to debug this). Basically the phones are able to place calls but not to receive calls. The extension is ringing for the calling party but the handsets do not ring. By putting the IP600 in debug mode and

...tofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <phoneLabel>Test2</phoneLabel> <stutterMsgWaiting>2</stutterMsgWaiting> <callStats>false</callStats> <offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig> <startMediaPort>16384<...

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

...Mail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>false</natEnabled> <natAddress></natAddress> <phoneLabel>Alker Study</phoneLabel> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>true</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</st...

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .New triggered check added: 1 [Dec 12 15:39:19]

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is