similar to: How to use TRUNK only if IAX fails?

Many Thanks Carlos, I was hoping to check whether the remote server is available before I issue the dial in my dial plan. Is there a better way to do it in asterisk without using unix commands? Many Thanks, Ashwin On 5/30/15, 2:06 AM, "Carlos Chavez" <cursor at telecomabmex.com> wrote: >On 5/29/15 1:16 PM, Ashwin Surendran wrote: >>> Hi, >> I have multiple

Hi Matt, I was a bit concerned on the delay if there might be any when my iax link is down? It would be two dial steps right when my iax link is down. But I?m more than happy to try. Many Thanks, Ashwin. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt Riddell (lists) Sent: 30 May 2015 16:55 To: Asterisk Users Mailing List -

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

I am currently having a voice quality problem with one of our Asterisk servers. We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions. I am looking at the timing source for Asterisk and it is currently using timerfd even though we have an E1 card installed. Is timerfd better than dahdi? Any recommendations to

We've been having some problems with an E1 PRI line for a few days. We get the following errors: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 INVOKE ID: 316 [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 PROBLEM: Invoke: Unrecognized Operation The telephone company says that

The command he gave you was in Asterisk. Why do you not want to call it to try it? Then you can fail over to the other trunk if the IAX link is down. Kind regards, Matt > On May 30, 2015, at 2:03 AM, Ashwin Surendran <Ashwin.Surendran at now-health.com> wrote: > > Many Thanks Carlos, I was hoping to check whether the remote server is > available before I issue the dial in

I would especially look at the CHANUNAVAIL dial status Since it sounds like you are probably qualifying your IAX trunk, that status will be the quickest way to overflow from IAX to TDM. On Sat, May 30, 2015, 11:35 PM Ashwin Surendran < Ashwin.Surendran at now-health.com> wrote: > Hi Matt, > > > > I was a bit concerned on the delay if there might be any when my iax link

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.

All; I have a customer who is looking for a good speech to text solution, either open source or reasonably priced commercial product, I'm open to suggestions. Thanks; John V -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150826/64d6c224/attachment.html>

hello list, i use chanspy with the code below [app-chanspy] exten => _007.,1,Macro(user-callerid,) exten => _007.,n,Answer exten => _007.,n,Authenticate(1111) exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use

Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the following error now: Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 12 14:42:35] WARNING[24498]:

Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf => mysql,asterisk,bit_ast_config - The following is the table in the database: mysql> select * from

I just purchased an Amfeltec USB-FXO adapter and am trying to compile DAHDI 2.10 on a Raspberry PI running Pidora 2014 R3. I have all the dependencies but I get an error and cannot finish. Is it even possible to compile DAHDI for the ARM plataform? Here is the error I am getting: root at astpi dahdi-linux-complete-2.10.2+2.10.2]# make make -C linux all make[1]: Entering directory

On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: > >> Has anyone successfully used Mysql realtime PJSIP with Asterisk >> 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the >> following error now: >>

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Since a couple versions back I keep getting these messages when compiling Dahdi: make[2]: Entering directory `/usr/src/kernels/3.10.0-327.13.1.el7.x86_64' INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_dynamic.ko Can't read private key

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]:

I am unable to force a hangup on a channel that has been stuck for over two days: IAX2/from-CD-11006 oficina 2770 1 Up Dial IAX2/to-CD/2883 3467130007 46:24:59 Sotelo Sotelo IAX2/to-CD-20713 I have tried "hangup request IAX2/from-CD-11006" several times but no joy. I also see the following in the CLI: [Nov 3

Ok, commented out that line. It's still doing it. Reloaded dialplan. Please don't tell me I have to restart asterisk. Thomas M. Peters | Systems Administrator | tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org >>> Tiago Geada <tiago.geada at gmail.com> 6/26/2015 12:07 PM >>> messages => error states to log error messages to