similar to: Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?

Hello. Who can add asterisk16-xmpp module to packages.asterisk.org or build asterisk with support xmpp and update packages? Thank You. -- Vasiliy G Tolstov <v.tolstov at selfip.ru> Selfip.Ru

Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

Le 15/01/2020 à 19:24, Administrator a écrit : > Hi all, > > we face a strange behavior while connecting an Asterisk16 instance > with PJSIP to 2 providers: we receive error 401 Unauthorized, both of > them having Kamailio as front-end. With other providers -we don't know > if they run kamailio- registration is just fine. > > One of the provider took a pcap and told

On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net> wrote: > > Le 17/01/2020 à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange behavior while connecting an Asterisk16 instance > >> with PJSIP to 2 providers: we receive error 401 Unauthorized,

Hi everybody, First of all i am sorry my English :) i want to configure my asterisk server as a sip server that stores sip users in the mysql database connecting directly over odbc driver. My odbc configuration works as below [root at ao042 asterisk]# isql -v asterisk +---------------------------------------+ | Connected! | | | |

Le 19/01/2020 à 00:31, Joshua C. Colp a écrit : > On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > > Le 17/01/2020 à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

Hello, does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our callerID is no more seen by them. We use Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or equal to CALLERID(num). We tried

>Information on timing sources can be found here: >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces >As noted on that page, ConfBridge can use any timing interface Asterisk >provides, and is not limited to the DAHDI timing interface. Generally, >timerfd is a good timing interface. >That aside, I would try to rule out external issues with the garbled audio

hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using "res_timing_dahdi" or I can use "res_timing_timerfd" to get some benefit if I upgrade to 1.8? thank a lot for

starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these "timing" modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

For those who are using SuSE: At last they've managed to create ready-to-run packages for openSUSE_11.1. They are there since a couple of hours... (For other versions it was allready available for some time on the OBS) /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/asterisk16-devel-1.6.0.6-82.2.x86_64.rpm

Hi Guys, I have tested and isntalled Asterisk 1.6.2 with FreePBX from Digium repos based on this url: http://www.asterisk.org/downloads/yum BUT that doesn't seem to work with Fedora instance which I am running on Amazon Ec2. Apparently Asterisk 1.4 is natively included in Fedora repository but not Asterisk 1.6. And when I added the Digium repository, it give me a 404 not found. I check and

On 2020-01-15 11:24, Administrator wrote: 8<'s > One of the provider took a pcap and told us that expiration was set to 0 > that's why they don't accept the registration. We took a pcap on our > side when SIP packet goes out of our server and we see that the > expiration parameter is setted to 3600 ! Howdy, Maybe the clipping of your SIP packet is occurring on