search for: asterisk16

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

...ory=/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 : asterisk16*CLI> moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh I've done the following : asterisk16*CLI> module reload res_musiconhold.so asterisk16*CLI> moh reload asterisk16*CLI> core restart now How to load new musiconhold classes ?? Using as...

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

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 /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/libspandsp1-0.0.5pre4-59.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/zaptel-kmp-default-1.4.12.1_2.6.27.19_3.2-32.2.x86_64.rpm /srv/distro/repo/network:/telephony:/as...

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 us that expiration was set...

...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, 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...

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 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 kamailio- registration is just fine. >> >> One of the provider took a pcap and told us that...

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

On Sun, Jan 19, 2020 at 10:45 AM Administrator <admin at tootai.net> wrote: <snip> > It become stranger and stranger: on one of the register peer we receive in > asterisk: > > *CLI> [2020-01-19 15:23:18] WARNING[17469]: > res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal > response '401' received from

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

...gt; <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 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. > >> > >&g...

Le 15/01/2020 à 19:50, C.Maj a écrit : > 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

Hi Everyone, I have done "yum install speex libspeex-devel speex-devel" and it was succesful on CentOS. I then tried "yum install asterisk16 asterisk16-addons asterisk16-configs" but "core show translation" doesn't show speex loaded. Is there a way to or an option that I can append to the asterisk install to make sure it compiles with speex in mind? Thanks, Bruce -------------- next part -------------- An HTML attach...

...ing 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: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 .

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 replacing + with 00, same problem. There support said they don't receive any callerID and we know that it's not correct as we don'...

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

...hen 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 if someone is using successfully a PJSIP channel against Kamailio. Another one: despite the fact that they use 5061 port, it's not TLS but UDP. Our asterisk16 has no TLS configured. We use wizard which looks like: [Provider-tootai](!) ; type = wizard sends_auth = yes sends_registrations = yes accepts_auth = no accepts_registrations = no endpoint/call_group = 1 endpoint/pickup_group = 1 endpoint/accountcode = TOOTAi endpoint/language = fr endpoint/allow...

...Nat ACL Port Status Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] besides it does not query anything. So what am i missing? Is there anything that i should mentioned in the sip.conf file? by the way i am using RHEL 5.4 with 2.6.18-164.el5 kernel - asterisk16-1.6.0.17-1_centos5 rpm Any help would be appreciated... -- Emre Kurnaz ITU/BIDB | Istanbul Technical University / Information Technologies Office Sistem Destek Grubu | System Support Team RHCE : 805009174841679 Yar? Zamanl? ??renc...

This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it doesn't yet exist, what is the process for upgrading?