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...obal] 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.connected: Connected Peer2.othernames: (Then volume options are listed) --- Volume configuration: root at int-gluster-03:~ # gluster volume info my_volume_name Volum...

...anuary 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" peer2@192.168.1.1 (where the...

...ions] >> 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.connected: Connected >> Peer2.othernames: >> >> (Then volume options are listed) >> >> >> --- >> &g...

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:

...-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.connected: Connected > Peer2.othernames: > > (Then volume options are listed) > > > --- > > > Volume configuration: > > root...

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" peer2@192.168.1.1 (where the...

...anuary 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" peer2@192.168.1.1 (where the...

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.

...ions] >> 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.connected: Connected >> Peer2.othernames: >> >> (Then volume options are listed) >> >> >> --- >> &g...

...2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On peer2: [dundi-incoming] exten => _X.,1,NoOp(Received EXTEN ${EXTEN}.) exten => _X.,n,Set(EXTTODIAL=${CUT(EXTEN|#|1)}) exten => _X.,n,Set(DUNDIVAR=${CUT(EXTEN|#|2)}) exten => _X.,1,NoOp(Extracted extension ${EXTTODIAL} and DUNDi variable ${DUNDIVAR}) exten => _X.,n,Goto(local-extensions,${...

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

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

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" error? Next idea is to use both peers i...

-- 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?

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes jitterbuffer=no disallow=all ;allow=ulaw ;allow=alaw allow=h261 allow=h263 allow=gsm -- IAX2/192.168.0.7:4569/2 answered SIP/1102-62b6 Sep 23 11:49:33 DEBUG[1099414448]: chan_sip.c:825...

Hey Guys I have an indial range of 61211118[01234]X being trunked sip to xxx.yyy.189.65 Now I want to break this down into 612111180x going to xxx.yyy.188.145 and 612111184x going to xxx.yyy.189.199 reminder being used for fax->email etc etc etc I have created the outbound routes and sip trunks I can see that all the sip trunks are up I can see the outbound routes are there and

If I set a call-limit field on a peer in users.conf.. I am seeing that it seems to affect other peers too? I am running Asterisk 1.4.18 ....has someone seen this issue. Peer 1 has call-limit=5 Peer 2 has call-limit=20... In the SIP trace I see when Peer2 hits 5, Asterisk sends back a 480 (temp Unavailable (Call limit reached)...msg.. Any ideas would be appreciated Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/580088f4/attachment.htm

Hi Everyone, In order to make life easier and to do debugging easier I want to observe "sip set debug originator" and "sip set debug terminator" on two different putty screens. Trick is that originator calls the terminator. I can of course put two separate calls and get sip debugs at different times but that's not what I want to do. I want both to spit out on my two