similar to: SIP debugging enabled : written to log ??

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

Would it be possible to execute some kind of script when for example Asterisk restarts... or stops... ? How can one read the status of Asterisk so that when the service is stopped I could be notified by mail, by text message,... ? I don't know how to read the status of Asterisk (or the change of status) in a bash-script. Thanks for the reply ! Kind regards, Jonas. -------------- next part

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able

Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: > Hi, > >

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

>From voip-info.org : Asterisk will look for these files in the /var/lib/asterisk/keys directory, so copy them there and make sure only the asterisk user id can read the keys and that no one can write over them. How do I know my "Asterisk User ID" ?? Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 --> 5064 is open. It's just very strange that some can register and other not. Any

It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--------------------------+--------+ | Variable_name            | Value  |

Hello list, google returns a discussion on the dev-list when I search for how to mail a voicemail to multiple mail addresses. Is there yet a seperator that actually works to define multiple mail addresses ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

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 list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hi, Just a question. I have a samba PDC with LDAP backend. I want squid / dansguardian use the user auth from samba NTLM bases. i need user and group filtering and i want it transparent. 2 steps, auth, first the NTLM auth on port 80 to be transparent. second, the dansgadian filter filtering groups. this looks bit like it, but this authenicates against ADS.

Hello, I have Snom, Cisco, Grandstream & YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101202/ea25cd7e/attachment.htm

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, are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ?