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...seems a very interesting tool. Thanks for sharing this. Now I'm still curious to learn if moving from local file storage to cloud IMAP storage is still resilient to short network outages (without IMAP replication). Regards 2015-05-06 10:18 GMT+02:00 Gareth Blades <mailinglist+asterisk at dns99.co.uk>: > On 05/05/15 17:52, Olivier wrote: > >> 2. From personal experience, would you rate an IMAP migration as an easy >> or as a difficult task ? >> By IMAP migration, I mean changing from one IMAP software to another, on >> the same or on an other box. >&gt...

Hi, We've recently hit an issue with Asterisk 18.8.0 where a call comes in via SIP (using pjsip) but it can take 5 seconds before starting to execute the dialplan. This was intermittent, but frequent (eg approx half of the calls). We have verbose logging on, but I didn't see any errors. Running asterisk -r -vvvv and then watching SIP traffic in another window showed the INVITE coming

We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone.

Hello, I'm currently studying what is needed to implement IMAP Voicemail with Asterisk 11 and up. More precisely, I would like to let users check voicemail with their smartphone from outside (ie not connected to LAN). My first questions are: 1. What happens if Asterisk cannot reach its configured IMAP store ? Are voicemails locally stored in a persistent directory surviving reboots or are

Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

...xMonitor Recording SIP/300-00000192 [2015-03-27 18:35:28] WARNING[18275]: chan_sip.c:23527 handle_response_register: Got 423 Interval too brief for service fdmaroc2 at sip.serveurcom.com, minimum is 480 seconds thanks nd regards 2015-03-27 17:08 GMT+00:00 Gareth Blades <mailinglist+asterisk at dns99.co.uk>: > You would need to give more information really. > Your sip.conf file listing the entries for the phones especially which > codecs are permitted. > A copy of the 'asterisk -rvvv' console output when you make the call. > > > > On 27/03/15 17:05, Salahedd...

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >

On 06/02/15 07:54, Olli Heiskanen wrote: > My goal is to allow my users record their own queue announcements and > choose which announcements they want to use in each queue. I have > several Asterisk servers and a Kamailio server which dispatches call > traffic between the Asterisks. Question is, is it possible to have > something like a NSF disk shared between several asterisk

On 05/05/15 17:52, Olivier wrote: > 2. From personal experience, would you rate an IMAP migration as an > easy or as a difficult task ? > By IMAP migration, I mean changing from one IMAP software to another, > on the same or on an other box. There is software called 'imapsync' which will sync mail from one imap store over to another. I have used that to migrate from one

We have a customer reporting poor quality calls when they come to us via a particular provider. The SIP traces look perfectly normal both on the ingress to us and egress to another telco. No additional sip messages after the call has been answered until the BYE is received. However in the asterisk logs I am getting this :- 2014-02-05 13:45:03 | C-00108c80] rtp_engine.c: -- Locally bridging

Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"

We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488