search for: voip2

...ies to route a call 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...

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 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...

...k 5 [Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio format found to offer. Cancelling call to 1XXXXXX4332 -- Couldn't call t564/1XXXXXX332 == Everyone is busy/congested at this time (0:0/0/0) I've checked to ensure that both formats are loaded into Asterisk: voip2*CLI> module show like 729 Module Description Use Count format_g729.so Raw G729 data 0 1 modules loaded voip2*CLI> module show like 723 Module Description...

...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 the tandem and gets dumped to vmail, does it go VOIP2<->VOIP1<->VMAIL1 or VOIP2<->VMAIL1? When user is talking on PSTN over Teliax, I can see TANDEM1 doing the transcoding if necessary and bridging via "IAX2 show peers". This leads me to believe it...

...I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be> Remote-Party-ID: "VC" <sip:voip2 at sip.domain.be>;screen=yes;party=calling Call-ID: 307124bd-f6881ef at 192.168.1.106 CSeq: 101 INVITE Max-Forwards: 70 Contact: "VC" <sip:voip2 at 192.168...

...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

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

...Num(${DEFAULTCID}) exten => s,5,SetVar(GATEWAY=${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(dialout,IAX2/voip1/${ARG1}) exten => s,15,GotoIf($["${ARG3}"...

...r Pooled: no Connected: yes Reference URLs for func_odbc are: http://www.asterisk.org/func_odbc http://svncommunity.digium.com/view/func_odbc/1.2/ The error I'm getting is: Jul 10 12:07:04 VERBOSE[30281] logger.c: -- Executing Set("SIP/4053-b4410638" , "ODBC_ASTDB_CLUSTER(voip2|CF/4053)=NULL") in new stack Jul 10 12:07:04 DEBUG[30281] pbx.c: Function result is 'voip2' Jul 10 12:07:04 DEBUG[30281] pbx.c: Function result is 'CF/4053' Jul 10 12:07:04 ERROR[30281] func_odbc.c: Unable to load ODBC write class (check res_odbc.conf) func_odbc.conf: [ASTD...

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 account details are t...

...:6688 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers Apr 27 11:33:53 WARNING[16384]: chan_iax2.c:6099 set_config: Ignoring port for n ow Apr 27 11:33:53 WARNING[16384]: chan_iax2.c:5928 build_user: Unable to support t runking on user 'voip2' without zaptel timing Apr 27 11:33:53 WARNING[16384]: chan_iax2.c:5756 build_peer: Unable to support t runking on peer 'voip2' without zaptel timing == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' ______________________...

...imple-message-summary Content-Length: 88 Messages-Waiting: no Message-Account: sip:102 at xxx.xxx.xxx.xxx Voice-Message: 0/0 (0/0) <--- SIP read from y.y.y.y:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0e58862b;rport From: "anonymous" <sip:anonymous at voip2.makaicomm.com>;tag=as69473f09 To: <sip:sip:102-Acme at 192.168.1.4:5060>;tag=D888A873-3AA22F98 CSeq: 112 NOTIFY Call-ID: 2b46ccef-d87e0939-6db3c26 at x.x.x.x Contact: <sip:102-Acme at x.x.x.x:33475> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0130 Content...

Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: 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 creat...

...rbd1: Local backing block device frozen? [242682.016071] block drbd1: Local backing block device frozen? [242694.048071] block drbd1: Local backing block device frozen? [242706.080071] block drbd1: Local backing block device frozen? [242718.112077] block drbd1: Local backing block device frozen? sb-voip2@sbvoip2:~$ sudo cat /proc/drbd version: 8.3.11 (api:88/proto:86-96) GIT-hash: 0de839cee13a4160eed6037c4bddd066645e23c5 build by root@sbvoip2, 2013-02-19 08:30:51 1: cs:Connected ro:Primary/Primary ds:UpToDate/UpToDate D r----- ns:14732 nr:1784712 dw:1799444 dr:579340 al:31 bm:44 lo:1 pe:0 ua:...

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to voicemail for that extension via Asterisk. My interim goal is to take any call coming in via the land line and immediately drop it into my main menu. I've got PSTN-to-VoIP Gateway enabled, w...

...(Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers => mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI> realtime mysql status localhost configured for mya2billing at localhost, port 3306 with username a2billinguser. mya2billing configured for mya2billing at localhost, port 3306 with username a2billinguser. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime...

....112> Content-Type: application/sdp Content-Length: 243 v=0 o=root 12813 12813 IN IP4 10.215.147.112 s=session c=IN IP4 10.215.147.112 t=0 0 m=audio 13290 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- inf-voip2*CLI> <--- SIP read from 10.215.146.162:5060 ---> ACK sip:4053 at 10.215.147.112 SIP/2.0 Via: SIP/2.0/UDP 10.215.146.162:5060;branch=z9hG4bKa6f748e3affb009b From: "TEST" <sip:4062 at pbx.voip.local>;tag=23bfef509d1f572f To: <sip:4053 at pbx.voip.local>;tag=as0c4f99e6 C...