similar to: Directory app not working with realtime

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function 'transport_apply': res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-00000074): Buffer overflow detected (59 + 134 > 175) Has anyone ever seen this before? I have the following configuration. udptl.conf:

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2, and 12.3.2. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases These releases resolve security

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2, and 12.3.2. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases These releases resolve security

Is there a way to tell if a particular fax was transmitted by T.38 or G.711? I'd like to save the information in the CDR, but the "fax show stats" command only gives a summary since the last restart. It would be nice if the FAXOPT variables contained this information but it doesn't. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax)

All; I'm working on a project (using Asterisk 1.8, but 11 would probably work just as well) where so far I've been able to originate over 1,000 concurrent outbound faxes. I have no problem with that so far. Where I have the problem is that Asterisk is dumping core after the faxes are sent. Now two things happen after the faxes are sent. (1) A fax log similar to a CDR is written to

All; I have a few customers that do a lot of faxing, both inbound and outbound. Some use the Spandsp and some use the Digium FFA modules. What I would like to do is when an outbound fax fails, determine whether the remote end was a fax machine or a plain old phone. I would also like to determine that in the dial plan if I could. Any insight at all with this would be extremely helpful.

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

All; I have a customer who is looking for a particular DID. (I dialed it and it is not in service). I searched through my preferred upstream provider's list but I came up empty. I wrote them, and this is their reply. "We currently do not have that specific number in stock as this number is owned by another carrier that we do not have a business relationship with." So my

hello and thank you so much for your reply just one question: how do you use it? AFAIK, when asterisk receive a call, it select extensions based on the received number and when it detects fax, jump to the fax extension. now when should i use these show commands to detect fax and how should i tell asterisk to execute fax extension? this is a big problem for me, so i really appreciate if you help

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

The Asterisk Development Team would like to announce the release of Certified asterisk-18.9-cert7. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7 and https://downloads.asterisk.org/pub/telephony/certified-asterisk This release resolves issues reported by the community and would have not been possible

The Asterisk Development Team would like to announce the release of Certified asterisk-18.9-cert7. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7 and https://downloads.asterisk.org/pub/telephony/certified-asterisk This release resolves issues reported by the community and would have not been possible

Hi I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. Connection to Asterisk all are based on SIP and SIP Trunks so no DAHDI hardware is required. According to some recommendations like http://osdial.org/howto/? "Internal timing is very critical with Asterisk when it is under load" and we must use DAHDI hardware or "USB Voice

On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote: > use replication 1. Agreed - use replication. 2. If you want an HA (High Availability, not dependent on a single Master DB server replicating to a slave) solution, consider setting up Master-Master replication, with an LVS (Linux Virtual Server) HA machine in front of the two, so that writes can go to either server using only

On 04/29/2017 10:57 AM, Jonathan H wrote: > On 29 April 2017 at 16:47, Tech Support <asterisk at voipbusiness.us> wrote: > >> I?m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. However, when I try to compile it, I?m getting hundreds and hundreds of errors. Here is a sample of the output. >> When I try to build Asterisk 13, I have no problem. Any

2015-05-06 17:51 GMT+02:00 Tech Support <asterisk at voipbusiness.us>: > Hey; > > It seems to me that for what you want to do, it would be easier just > to email the user the voicemail audio file as an attachment. > Yes that's true but still, for users with an "intensive voicemail usage", unified messaging is attractive. In the long run, is unified

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.8.11-cert7, 1.8.15.1,

The Asterisk Development Team has announced the release of Certified Asterisk 13.1-cert7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.1-cert7 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the