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...n E5-2660 with 16GB RAM virtualised on XenServer 7.2 root at int-gluster-03:~ # gluster get glusterd state dumped to /var/run/gluster/glusterd_state_20170915_110532 [Global] MYUUID: 0b42ffb2-217a-4db6-96bf-cf304a0fa1ae op-version: 31200 [Global options] cluster.brick-multiplex: enable [Peers] Peer1.primary_hostname: int-gluster-02.fqdn.here Peer1.uuid: e614686d-0654-43c9-90ca-42bcbeda3255 Peer1.state: Peer in Cluster Peer1.connected: Connected Peer1.othernames: Peer2.primary_hostname: int-gluster-01.fqdn.here Peer2.uuid: 9b0c82ef-329d-4bd5-92fc-95e2e90204a6 Peer2.state: Peer in Cluster Peer2....

...> glusterd state dumped to /var/run/gluster/glusterd_state_20170915_110532 >> >> [Global] >> MYUUID: 0b42ffb2-217a-4db6-96bf-cf304a0fa1ae >> op-version: 31200 >> >> [Global options] >> cluster.brick-multiplex: enable >> >> [Peers] >> Peer1.primary_hostname: int-gluster-02.fqdn.here >> Peer1.uuid: e614686d-0654-43c9-90ca-42bcbeda3255 >> Peer1.state: Peer in Cluster >> Peer1.connected: Connected >> Peer1.othernames: >> Peer2.primary_hostname: int-gluster-01.fqdn.here >> Peer2.uuid: 9b0c82ef-329d-4bd5...

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...t; root at int-gluster-03:~ # gluster get > glusterd state dumped to /var/run/gluster/glusterd_state_20170915_110532 > > [Global] > MYUUID: 0b42ffb2-217a-4db6-96bf-cf304a0fa1ae > op-version: 31200 > > [Global options] > cluster.brick-multiplex: enable > > [Peers] > Peer1.primary_hostname: int-gluster-02.fqdn.here > Peer1.uuid: e614686d-0654-43c9-90ca-42bcbeda3255 > Peer1.state: Peer in Cluster > Peer1.connected: Connected > Peer1.othernames: > Peer2.primary_hostname: int-gluster-01.fqdn.here > Peer2.uuid: 9b0c82ef-329d-4bd5-92fc-95e2e90204a6 >...

Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/number at domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: number at domain2.com@domain1.com I need: number at domain2.com I can't use "SIP uri dial", i need authorization (peer1) I think asterisk can't do that. Is w...

...ant to configure the client asterisk with the server asterisk as IAX2 peer and want to connect to the IPV6 ip. I bind the server with ipv6 and also sending the registration request from the client(peer) to the ipv6 address. But its not peering. following is the client's iax.conf register => peer1:peer1pass@[IPV6]:port [peer1] type=peer context=topeer username=peer1 secret=peer1pass trunk=yes host=XXX.XXX.XXX.XXX port=XXXX disallow=all allow=g729:40,g723:30 qualify=yes Also my confusion is what value will be in 'host' property. I assigned as host=[IPV6]...but it shows error. Can an...

...> glusterd state dumped to /var/run/gluster/glusterd_state_20170915_110532 >> >> [Global] >> MYUUID: 0b42ffb2-217a-4db6-96bf-cf304a0fa1ae >> op-version: 31200 >> >> [Global options] >> cluster.brick-multiplex: enable >> >> [Peers] >> Peer1.primary_hostname: int-gluster-02.fqdn.here >> Peer1.uuid: e614686d-0654-43c9-90ca-42bcbeda3255 >> Peer1.state: Peer in Cluster >> Peer1.connected: Connected >> Peer1.othernames: >> Peer2.primary_hostname: int-gluster-01.fqdn.here >> Peer2.uuid: 9b0c82ef-329d-4bd5...

We're using Asterisk 14.7.6 and I have a dialplan that ends like this: same => n,Dial(SIP/${EXTEN:0:4}@peer1) same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) same => n,Hangup() When peer1 hangsup, the priorities after the Dial are executed fine. But when the caller hangsup during the Dial, the cleanup steps aren't done. Why? I did read "Note that on a successful connection, in the absenc...

...lf Of Damon Estep Sent: Thursday, January 04, 2007 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk sip peer/user matching methods forauthentication backwards? Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 ... [peer2] Host=dynamic Secret=password ... [Peer3] Config not relevant ... The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes "From...

I have been working through the book "Applied longitudinal data analysis: modeling change and event occurrence" by Judith D. Singer and John B. Willett. I have been working examples using SAS and also using it as an opportunity for learning to use R for statistical analysis. I ran into some difficulties in chapter 8 which deals with using structural equation modeling. I have tried to

...e, Jun 5, 2018 at 1:21 AM, Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Tuesday 05 June 2018 at 08:33:26, David P wrote: > > > We're using Asterisk 14.7.6 and I have a dialplan that ends like this: > > > > same => n,Dial(SIP/${EXTEN:0:4}@peer1) > > same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) > > same => n,Hangup() > > > > When peer1 hangsup, the priorities after the Dial are executed fine. But > > when the caller hangsup during the Dial, the cleanup steps aren't done. > > Why? > >...

Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 ... [peer2] Host=dynamic Secret=password ... [Peer3] Config not relevant ... The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes "From...

...lf Of Damon Estep Sent: Thursday, January 04, 2007 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk sip peer/user matching methods forauthentication backwards? Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 ... [peer2] Host=dynamic Secret=password ... [Peer3] Config not relevant ... The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes "From...

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not answering (in 3sec) or send "50x"...

Hello I have two peers setup and working with x2 bricks each. They have been working via tcp for the last 4-5 months. I just got two Infiniband cards and put the on the peers. I want to change the transport type to rdma instead of tcp but I don't see an easy way to do this. Can you please help me with proper instructions. Best Regards Ivan Dimitrov

http://bugzilla.mindrot.org/show_bug.cgi?id=1282 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution| |WORKSFORME CC|

...at asterisk. >> open.source.it>> wrote: >> >> On Tuesday 05 June 2018 at 08:33:26, David P wrote: >> >> > We're using Asterisk 14.7.6 and I have a dialplan that ends like >> this: >> > > same => n,Dial(SIP/${EXTEN:0:4}@peer1) >> > same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) >> > same => n,Hangup() >> > > When peer1 hangsup, the priorities after the Dial are >> executed fine. But >> > when the caller hangsup during the Dial, the cleanup steps ar...

...rescue64.c32 and maybe a few others. Basically, it is ignoring the rescue64 label and trying to directly spit a rescue64 file out at me. I can see this from the tftpd logs. Has anyone else seen similar behaviour? Any tips or ideas? Greatly appreciated, John -- John Newman Product Engineering - Peer1 Hosting

-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how?

Hi, all. Is there any chance to set individual CALLERID(num) for channels SIP/peer1, SIP/peer2 in a call Dial(SIP/peer1&SIP/peer2). There is an option to use Dial(SIP/peer1&SIP/peer2,,M(set_callerid)), but the macro will be launched after the channel answered. Not really want to use local channel because of not quite usable cdr. Thanks.