similar to: Hold

Hi ?? I have been trying for several days get 3 Cisco spa508g phones (firmware 7.5.5) to work with asterisk 11.6 cert1 and sla. I can get the phones to all ring when an incoming call arrives, and I see the slatrunk working. However the blf function does not work. If one extension picks up the call the others do not show the trunk in use.? And as you might expect the hold and outbound dialing does

Hi. Little story :-) I'm playing with dovecot 2.2.25 and multiple SSL certificates. ~7000 certificates which are loaded twice, so my dovecot has ~14 000 certificate pairs (14k key + 14k cert) in config. 14 000 local_name entries. Like these: local_name imap.example.com { ssl_cert = </etc/certs/cert1.pem ssl_key = </etc/certs/cert1.pem } local_name pop3.example.com { ssl_cert =

Dear list, i have a issue Asterisk crash (Module res_odbc exactly) after the same log who is "/ERROR[23805] astobj2.c: bad magic number.../" you will see on the log : Today [2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number 0x552f302e for 0x7fe3505b3958* [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on a x86_64 running Linux on 2013-04-09

On 10-09-16 00:50, Richard Mudgett wrote: > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list > of my queues and then the following : > > > failed to extend from 240 to 327

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen queue -> Someone picks up the phone (Internal ext. 321) -> CallerID shown on customers

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc77 at 178.19.90.240>' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]:

The Asterisk Development Team is pleased to announce the first release of Certified Asterisk 11.6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-11.6-cert1 Thank you for your

Hello list, i run here an large mailsetup with some million mailboxes and got strange performance problems, cause i think i have overseen or forgotten an simple setting. Here are some details: 21 CentOS 7 Servers with dovecot 2.2.25 and ldap userdb/passdb via socket behind an hardware loadbalancer. The storage behind is an ISCSI Storage with 4 10Gbit/s multipath paths, splitted up to 10 TB

Hello list, when reloading dovecot via "doveadm reload" it throws me tons of errors like this: Feb 08 10:57:30 server1 dovecot[18243]: Feb 08 10:57:30 imap: Error: net_connect_unix(/run/dovecot/auth-master) failed: Resource temporarily unavailable Feb 08 10:57:30 server1 dovecot[18243]: Feb 08 10:57:30 imap: Error: net_connect_unix(/run/dovecot/auth-master) failed: Resource temporarily

Hi Is there any way to park a call on an extension. Say a customer calls in and wants to speak with Steve. Steve is currently on the phone. How can I park the call so that when Steve hangs up his phone immediately rings and he is connected to the parked call. Thanks Kelly

The Asterisk Development Team is pleased to announce the release of Certified Asterisk 13.1-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-13.1-cert1 Thank you for your

Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem.

When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]]) options r: Send ringing

(Sorry if this has hit the list before. I sent it yesterday, but never saw it come through) I actually got rid of my phone service.. no more pots line in my house. But, I miss my call screen macro. Any way to do this with a SIP channel? (Obviously the parking isn't the problem, but rather recording their name). I set it up so they should only have to record their name once provided they

Hello! We recently upgraded one of our customers from 1.4.44 to 1.8.15-cert1. We have several other customers running both versions. The customer in question does not use us as their provider as they?re located in a different country. When they make outgoing calls, there is a 3 second delay between answering the call and the call being established. When debugging this, I found that Asterisk

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

I have followed the instructions in Asterisk The Definitive Guide 4th edition. ?Once I load DAHDI I run the dahdi show channels command and get no such command. ?I have setup all the conf files. ?I compiled DADHI prior to Asterisk. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: