search for: voip1

...ll over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip address of voip1) port=5036 mask=255.255.255.255 qualify=yes ; Make sure this peer is alive trunk=no context=IAX ----------iax.conf on voip1----------- [voip2] type=friend username=voip2 host=x.x.x.x (ip address of voip2) port=5036 mask=255.255.255....

Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel Plugin Initialized Jul 15 11:43:57 voip1 pppd[9299]: Using zaptel device 'stdin' Jul 15 11:43:57 voip1 pppd[9...

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend host=10.0.0.160 auth=rsa secret= inkey=voip3 outkey=voip1 context=CONTEXT_VOIP1 allow=all ipaddr=10.0.0.160 port=4569 qualify=yes trunk=yes The iax.renoir.conf file contains : [VOIP_RENOIR] t...

...voip2*CLI> show uptime System uptime: 15 hours, 55 seconds voip2*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found voip2*CLI> show uptime System uptime: 15 hours, 1 minute, 28 seconds voip2*CLI> ----- Connected to Asterisk 1.2.27 currently running on voip1 (pid = 26496) -- Remote UNIX connection Verbosity is at least 3 voip1*CLI> show uptime System uptime: 4 days, 23 hours, 55 minutes, 1 second Last reload: 1 day, 4 minutes, 23 seconds voip1*CLI> ____________________________________________________________________________________ Y...

...o I force the vmail to go down the trunk to VMAIL1? D. How do I catch it on the other end and stick it only in a mailbox? Basically, how do I split the voicemail transfer off the local box to another? Now for a couple of architectural questions: 1. When a caller rings thru the TANDEM1 box to a VOIP1 extension, and then gets dumped to vmail, does the call go TANDEM1<->VOIP1<->VMAIL1 or does VOIP1 hand it off so it's only TANDEM1<->VMAIL1, presuming all IAX2 trunks are running a matching subset of codecs? 2. Same thing for intracompany calls. If VOIP2 calls VOIP1 user via...

...6 root root 4096 Feb 6 17:56 kernels drwxr-xr-x 2 root root 4096 Feb 9 23:19 libpri lrwxrwxrwx 1 root root 38 Feb 9 23:22 linux-2.6 -> /usr/src/kernels/2.6.9-42.0.8.EL-i686/ drwxr-xr-x 7 root root 4096 Feb 6 10:43 redhat drwxr-xr-x 10 root root 12288 Feb 9 23:25 zaptel [root@voip1 src]# cd /usr/lib/asterisk/modules/ [root@voip1 modules]# ls -l *zap* -rwxr-xr-x 1 root root 119069 Feb 9 23:26 app_zapateller.so That's the only thing there (with zap, that is). The zaptel compiled and installed ok, as I can run the zttool or ztcfg to see the cards being recognized and c...

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

...so login with user created with hylafax itself. But when I put auth required    pam_access.so auth            sufficient              pam_ldap.so account         sufficient              pam_ldap.so password        sufficient              pam_ldap.so in /etc/pam.d/hylafax, I get Jan 25 08:28:40 voip1 HylaFAX[1560]: pam_ldap(hylafax:auth): conversation failed Jan 25 08:28:40 voip1 HylaFAX[1560]: pam_ldap(hylafax:auth): conversation failed Jan 25 08:28:40 voip1 HylaFAX[1560]: pam_ldap(hylafax:auth): failed to get password: Authentication token manipulation error Same result with winbind and clas...

...ing at all. But, if i start asterisk with cli console (-vvvvvvc) theres is no problem and i can create the conference rooms. Zaptel and ztdummy modules are already loaded. Anyone with the same problem ? Asterisk: 1.0.8 Kernel: 2.6.9-5.0.3.ELsmp Regards, Jo?o Amaro ====> DUMP <==== [root@VoIP1 ~]# service asterisk restart [root@VoIP1 ~]# asterisk -vvvvvvvvvvvvr (...) ~ -- Executing MeetMe("OH323/R47", "1234|ciMps|") in new stack ~ == Parsing '/srv/etc/asterisk/meetme.conf': Found 2005-06-28 16:49:53 WARNING[4555]: channel.c:1913 ast_request: No channel typ...

...tension number configuration: transport=tls Finally, my phone was registered successfully on my asterisk server. But, during my tests and while I switched on sip debug mode, I have seen that on Register I have TLS and on Subscribe I have UDP. Please check the debug output bellow: 1. REGISTER: sip:voip1;transport=tls;lr SIP/2.0 Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:37156;rport;branch=z9hG4bKPjoCCw0.LEC-qhSMVBqFcWE8K4.jeEqwpI;alias Authorization: Digest username="2224", realm="asterisk", nonce="22603797", uri="sip:voip1;transport=tls;lr", response="125b4df128...

Hi, Sometimes my asterisk server stops. (after a day or two) Last output from CLI is: -------------------------------- -- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120 -- Channel 0/26, span 1 got hangup -- Hungup 'Zap/26-1' voip1*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). -------------------------------- I had tried both asterisk stable, 1.0.3 and several CVS-versions. Any clue how to solve this provblem ? By the way, if I do a 'restart now', asterisk on...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hi, I posted this also on hylafax list - maybe here is someone with a hint. System: Debian Jessie, Hylafax-Server 6.0.6, pam 1.1.8, libpam-ldapd 0.9.4, nslcd 0.9.4 (all actual debian packets from stable), sernet-samba-*-4.2.7-8 After a switch from OpenLDAP to a Samba 4.2 based LDAP Server, I cannot auth users anymore in Hylafax, everything else

Hi all, Today I got problem below and my domU become unresponsive and I should restart the pc to make it running properly again. [ 240.172092] INFO: task kworker/u:0:5 blocked for more than 120 seconds. [ 240.172110] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172376] INFO: task jbd2/xvda1-8:153 blocked for more than 120 seconds. [ 240.172388]

...istry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and remote IAX configurations. Local server: register => voip1p@voip2.test.net ; [voip2p] type=peer host=dynamic port=4569 trunk=no qualify=yes context=IAX Remote server: register => voip2p@voip1.test.net ; [voip1p] type=peer host=dynamic port=4569 trunk=no qualify=yes context=IAX

...3]: channel.c:1745 ast_request: No channel type registered for 'IAX' Apr 26 10:53:32 NOTICE[311313]: app_dial.c:554 dial_exec: Unable to create channel of type 'IAX' Apr 26 10:53:42 WARNING[311313]: pbx.c:1858 ast_pbx_run: Timeout, but no rule 't' in context 'sip' voip1*CLI> Any info would be appreciated. Thanks. My configs are below for sip, iax, and extensions viop1: sip-------------- [general] port = 5060 ; Port to bind to (SIP is 5060) context=sip ;inside_net=192.168.1.15 ;inside_mask=255.255.255.0 ;outside_addr=65.243.233.251 ;externip=65....

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

...> s,2,SetCIDNum(${ARG2}) exten => s,3,Goto(5) exten => s,4,SetCIDNum(${DEFAULTCID}) exten => s,5,SetVar(GATEWAY=${ARG3}) exten => s,6,SetVar(ARG3=${ARG4}) exten => s,7,SetVar(ARG4=${ARG5}) exten => s,8,SetVar(ARG5=${ARG6}) exten => s,9,GotoIf($["${GATEWAY}" = "voip1"]?14) exten => s,10,GotoIf($["${GATEWAY}" = "voip2"]?18) exten => s,11,GotoIf($["${GATEWAY}" = "voip3"]?16) exten => s,12,Macro(dialout,SIP/${ARG1}@pstn) exten => s,13,GotoIf($["${ARG3}" = ""]?20:5) exten => s,14,Macro(...

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it did not seem to have any problem. now I have placed another box to act as a firewall in front of the asterisk box and I can't seem to register both lines. the sip acc...

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at incoming_clients:1] Dial("SIP/toto.fr-28fdf000", "SIP/0825387205 at sipoperator") in new stack -- Called 0825387205 at sipoperator -- SIP/sipoperator-28fed000 is making progress pass...

...from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee type=friend host=dynamic secret=otherSecret context=local nat=no canreinvite=no qualify=no dtmfmode=rfc2833 allow=all call-limit=1 busy-level=1 allow=all [Caller] type=peer host=voip1.domain.net deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.xxx context=myAccess disallow=all allow=all nat=yes insecure=port,invite On the caller side [115] type=peer username=115 secret=blabla context=local host=dynamic nat=yes canreinvite=no dtmfmode=auto disallow=all allow=jpeg,png,h263,h263p,h264,al...