Re: [asterisk-users] Passing parameter to Queue-called macro

gt; The above works just fine for doing what I want to do, e. g. pass a > parameter from an Asterisk dialplan context into a queue-triggered "agent > just answered in the queue" Asterisk macro. > > Thanks for the reply! > > Kind regards > > Stefan > -Origi

Re: [asterisk-users] 7965G sporadically not able to make calls via chan_sip

Hi, the tcpdump starts with a pretty standard INVITE sequence: 10.0.0.121 -- INVITE --> 10.0.0.3 10.0.0.121 <-- 401 Unauthorized -- 10.0.0.3 // asterisk gives nonce in WWW-Authenticate: header 10.0.0.121 -- ACK --> 10.0.0.3 After that, normally you would see a new INVITE from the phone

Re: [asterisk-users] Passing parameter to Queue-called macro

Hi, maybe I am overlooking something, but channel variables should be thread safe, shouldn't they? I am using the following (sorry, in ael): macro dial-queue (number) { Set(_ORIG_UNIQUEID=${UNIQUEID}); Queue(${number},rCt,,,${timeout},,set-dst-agent); .. } // the

[asterisk-users] dnsmgr - lots of entries for the same host

Hello everybody, I am seeing a strange problem on Asterisk 1.8 with dnsmgr. The number of entries in DNS Manager seems to be growing steadily and all are pointing to the sama host - a SIP trunk to a local provider, which uses SRV lookup. So, when DNS manager refreshes, there are 6000+

Re: [asterisk-users] Update peer IP address

On 14.09.2015, at 21:58, Sebastian Kemper wrote: > So I got rid of the firewall rule that opened the RTP ports. And then it > dawned on me that I don't even need to open the 5060 port. The REGISTER > requests established a UDP connection that the kernel's conntrack module

Re: [asterisk-users] Strange Issue: asterisk deleted

On 26.11.2014, at 22:08, Antoine Megalla aa...@rocketmail.com wrote: The asterisk installation went fine but as soon as I start asterisk executable it loads everything and then after the Ready line the process gets killed and when I try to run it again i get: /usr/sbin/asterisk : command

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

On 22.11.2014, at 13:40, Yves A. yves...@gmx.de wrote: I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit.

Re: [asterisk-users] Voicemail message number off by one when using ODBC storage

... 'cause message file names start with 0 (msg.wav). -- marie On 05.10.2014, at 18:45, Leandro Dardini ldard...@gmail.com wrote: Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: Just wanted to let you know you were

Re: [asterisk-users] how can queue agents choose which call to answer?

calls correctly even when transferred early. On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter li...@mksolutions.info wrote: Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee: Hi everybody, I'm looking for a solution for the following scenario: • Asterisk queue

Re: [asterisk-users] how can queue agents choose which call to answer?

... and to continue my thought, if nothing else is possible, would it be a Very Bad Idea to just delete the ABANDON log (queue_log goes to mysql via odbc) automatically after it's created? In h extension? -- marie On 05.10.2014, at 20:42, Marie Fischer ma...@vtl.ee wrote: Thanks for your

[asterisk-users] how can queue agents choose which call to answer?

Hi everybody, I'm looking for a solution for the following scenario: • Asterisk queue • At peak hours, there will be more callers then queue members/agents, so some callers will spend some time on hold • Agents should be able to choose which of the on hold calls to answer instead of answering

Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

On 01.09.2014, at 11:42, Lukasz Sokol el.es...@gmail.com wrote: On 31/08/14 17:40, Marie Fischer wrote: Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit : we use OSX CardDAV server and its response is very slow, so we ended up syncing all the CardDAV contacts to MySQL via cron

Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

On 29.08.2014, at 22:44, Olivier oza.4...@gmail.com wrote: Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit : Hi, we use OSX CardDAV server and its response is very slow, so we ended up syncing all the CardDAV contacts to MySQL via cron. Asterisk dialplan then runs a query

Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

Hi, we use OSX CardDAV server and its response is very slow, so we ended up syncing all the CardDAV contacts to MySQL via cron. Asterisk dialplan then runs a query defined in func_odbc.conf. -- marie On 29.08.2014, at 15:03, Lukasz Sokol el.es...@gmail.com wrote: Hi, Would it be hard /

[asterisk-users] queue stats last reset date/time

Hello everybody, is there any way to find out when the queue stats ('queue show' / AMI action 'QueueStatus') was last reset (by 'queue reset stats')? These counters would make much more sense if I knew what timeframe they cover. ;) -- marie --

Re: [asterisk-users] Hangup cause 111 after call pickup

On 06.06.2013, at 15:05, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the

Re: [asterisk-users] Integration with skype

On 23.05.2013, at 12:57, bilal ghayyad bilmar...@yahoo.com wrote: There is no free channel to be used to have integration between asterisk and skype? What is the software that I can use to send and receive chat messages on skype network? For voice calls, you could try Skype Connect, which

Re: [asterisk-users] Stress testing Asterisk

On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote: Hi, I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer

Re: [asterisk-users] Stress testing Asterisk

is running and listen to some message played via Background(). - Forwarded Message - From: Marie Fischer ma...@vtl.ee To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 1:16 PM Subject: Re: [asterisk-users] Stress

Re: [asterisk-users] debug strategy for one-way audio calls

On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote: Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit : from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP

[asterisk-users] debug strategy for one-way audio calls

Hello everybody, from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some

Re: [asterisk-users] CDR unanswered setting

On 09.04.2013, at 10:33, Shanavaz E A shanava...@yahoo.com wrote: Hi, From asterisk 1.8, the CDR table is not logging the unanswered or extn busy calls which hit while in the queue. I am talking about this setting in the cdr.conf : ; In brief, this option controls the reporting

Re: [asterisk-users] Connect to an outbound channel and dial a phone number??

On 09.04.2013, at 23:12, Thomas Perron thomas.per...@gmail.com wrote: This seems basic but something is missing. I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number

Re: [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!

On 09.04.2013, at 23:43, Nick Khamis sym...@gmail.com wrote: That's just it! Nothing! It just does not pass the 91 mark. There are no failed calls during the test: Successful call|0 |20802 Failed call|0 |0

[asterisk-users] realtime peer w/ callbackextension does not register after 'sip reload'

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised

[asterisk-users] sip set debug on output to file only (not to console)

Hello everybody, I am trying to find an intermittent SIP error with one provider and thought the best first step would be to have sip set debug on for some days and check the logs. Everything gets logged nicely, but the SIP log clutters up the console quite badly. Is it possible to have SIP

Re: [asterisk-users] sip set debug on output to file only (not to console)

On 29.03.2013, at 15:05, Doug Lytle supp...@drdos.info wrote: Marie Fischer wrote: full = notice,warning,error,debug,verbose,dtmf,fax You should have a log called full in: /var/log/asterisk Sure I do and happy with that. :) The point is, I also have my Asterisk console full of SIP