search for: pbx3

...Op(**** RECEIVING FAX : ${FAXFILE} ****) exten => receive,n,ReceiveFAX(${FAXFILE}) exten => receive,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}" --cid-number ${CALLERID(num)} --cid-name "${CALLERID(name)}" --dest-name "Sir/Madam"') *debug:* pbx3*CLI> -- Executing [receive at fax-rx:20] ReceiveFAX("DAHDI/1-1", "/var/spool/asterisk/fax/2011-06-18_12:52:44_8009806858.tif") in new stack q931.c:5088 q931_connect: Call 14288 enters state 8 (Connect Request). Hold state: Idle pbx3*CLI> pbx3*CLI> > DL-DATA requ...

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't...

...ot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig so pkg-config --libs jack recognizes it. Modified /etc/ld.so.conf so libjack.so is cached in ld.so.cache: motiejus at pbx3:/etc$ strings ld.so.cache | grep jack libjackserver.so.0 /usr/local/lib64/libjackserver.so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-xmldoc,...

Hi List, I did follow the procedure to install Free Fax for Asterisk successfully till i came accross this isssue: i can't load the fax module: pbx3*CLI> module load res_fax_digium.so Unable to load module res_fax_digium.so Command 'module load res_fax_digium.so' failed. [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: und...

Does anyone know where I can find some good DUNDi docs? The ones are dundi.org are absolutely horrible. The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and

...nd 'B' (type 'help' for help) hera*CLI> C No such command 'C' (type 'help' for help) hera*CLI> D No such command 'D' (type 'help' for help) hera*CLI> E No such command 'E' (type 'help' for help) hera*CLI> [10:30][root@hera(pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Pub...

...tion? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608) attached which is generating a call, the other machine (box B) has the final destination. Sip config for the phone on box A (via Realtime): pbx3*CLI> sip show peer 2608 * Name : 2608 Secret : <Set> MD5Secret : <Not set> Context : assigned-device Language : AMA flags : Unknown CallingPres : Presentat...

...ping-table called-e164 to called-ip transfo map [0-9]+ to 192.168.100.254 mapping-table called-e164 to called-uri transfo2 interface isdn IF-PBX route call dest-table appels_provenance_ISDN interface isdn IF-PBX2 route call dest-table appels_provenance_ISDN interface isdn IF-PBX3 route call dest-table appels_provenance_ISDN interface isdn IF-PBX4 route call dest-table appels_provenance_ISDN interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table appels_vers_ISDN service sip-location-service ASTERISK_SRV bind location-se...

...map [0-9]+ to 192.168.100.254 > > mapping-table called-e164 to called-uri transfo2 > > interface isdn IF-PBX > route call dest-table appels_provenance_ISDN > > interface isdn IF-PBX2 > route call dest-table appels_provenance_ISDN > > interface isdn IF-PBX3 > route call dest-table appels_provenance_ISDN > > interface isdn IF-PBX4 > route call dest-table appels_provenance_ISDN > > interface sip IF-ASTERISK > bind context sip-gateway ASTERISK > route call dest-table appels_vers_ISDN > > service sip-loc...

...eues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx3,300,IAX,dundi3:${SECRET}@${IPADDR}/${NUMBER},nopartial extensions.conf(PBX1): [global_dundi_q_pbx1] include => one_queue_acd [global_dundi_q_pbx2] [global_dundi_q_pbx3] extensions.conf(PBX2): [global_dundi_q_pbx1] [global_dundi_q_pbx2] include => one_queue_acd [global_dundi_q_pbx3] ext...

I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends an ACK followed by the INVITE with the...

I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends an ACK followed by the INVITE with the...

...0 | | dundipbx2 | host | pbx2.ipt.XXX.com | 0 | | dundipbx2 | secret | FBoAsFz3S8hFQ | 0 | | dundipbx2 | type | peer | 0 | | dundipbx2 | username | dundipbx2 | 0 | | dundipbx3 | deny | 0.0.0.0/0.0.0.0 | 0 | | dundipbx3 | fromuser | dundipbx3 | 0 | | dundipbx3 | host | pbx3.ipt.XXX.com | 0 | | dundipbx3 | permit | xxx.yyy.142.201 | 0 | | dundipbx3 | permit...

...> pbx 2&3) > Exten => > 1XXX,1,Dial(IAX2/iaxtrunk:1234@XXX.187.142.203/${EXTEN}) > > ;To reach internal extensions on pbx2 (put this in > pbx 1&3) > Exten => > 2XXX,1,Dial(IAX2/iaxtrunk:1234@XXX.187.142.204/${EXTEN}) > > ;To reach internal extensions on pbx3 (put this in > pbx 1&2) > Exten => > 3XXX,1,Dial(IAX2/iaxtrunk:1234@XXX.187.142.234/${EXTEN}) > > > You could also specify each PBX in the [globals] > context > Example > > [globals] > > TRUNKPBX1 => IAX2/iaxtrunk:1234@XXX.187.142.203 > TRUNKPBX2...

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this

Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I