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...d=22545070 maxexpire=10 accountcode=roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. -------------- next part -------------- SIP Debugging Enabled Sip read: REGISTER sip:pbx SIP/2.0 Via: SIP/2.0/UDP 192.168.144.247 CSeq: 299 REGISTER To: "Roy Sigurd Karlsbakk" <sip:roy@pbx> Expires: 900 From: "Roy Sigurd Karlsbakk" <sip:roy@pbx> Call-ID: 1793030308@192.168.144.247 Content-Length: 0 User-Agent: KPhone/3.1 Event: registration Allow-Events:...

...update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk stasis.c:1490 publish_msg() # 6: [0x59588e] asterisk stasis_channels.c:796 ast_channel_publish_snapshot() # 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec() # 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper() # 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension() #10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec() #11: [0x53b599] asterisk pbx_app.c:493 pbx_exec() #12: [0x52e039] asterisk pbx.c:2989 pbx_e...

...ns failing. I'm guessing that they are dialling the wrong DTMF keys, OR that DTMF is hearing the digits entered wrong (or not hearing some). I've put debug => debug into logging.conf, and searched through the file, but I'm not sure how to debug. EG, Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '"USER ABC" <2060>' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2098' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'from-sip' Jun 1 14:32:33 DEB...

...[5296] netsock.c: == Using SIP RTP TOS bits 184 [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using SIP RTP CoS mark 5 [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using UDPTL TOS bits 184 [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using UDPTL CoS mark 5 [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [0000034635933565 at from- internal:1] Macro("SIP/21-00000058", "user-callerid,SKIPTTL,") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-user-callerid:1] Set("SIP/21-00000058", "AMPUSER=21") in new stack...

...sec,disposition,amaflags,accountcode) VALUES ('2006-02-09 11:00:30','x? ','x? ','0108680580','custom-did-route', 'Zap/2-1','SIP/580-2357','Dial','SIP/580|25|tr',5,2,'ANSWERED',3,'') Feb 9 11:00:35 DEBUG[25990] pbx.c: Function result is 'x? ' Feb 9 11:00:35 DEBUG[25990] pbx.c: Function result is 'x? ' Feb 9 11:00:35 DEBUG[25990] pbx.c: Function result is '0108680580' Feb 9 11:00:35 DEBUG[25990] pbx.c: Function result is 'custom-did-route' Feb 9 11:00:35 DEBUG[25990] pbx.c:...

...or all DS1s. How many D-Channels? If NFAS=Yes, Number of D-Channels Required "TWO" - (PRI ISDN Protocol)= example National PRI (NIPRI) NIPRI, NTNAPRI, U449PRI, U459PRI, N449PRI 3. Trunk Group Billing Telephone Number (10 digit): 4. Trunk Group Originating Area Code for PBX 5. Calling Scope (Basic, Basic+, Extended, Metro, Metro+)? 6. Primary Interexchange Carrier (PIC) 7. NPA/NXX Call Blocking ( 700, 900, 976) 8. Trunk Group Directional call flow and Member numbering: (If sub-grouped, select subgroup rollover) - 2-Way, 1-Way Into PBX, or 1-Way Ou...

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm really hoping so...

...ystem: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call* progress or busy detect. Also its * 1.2.13 w/ FreePBX2.2. I have scoured the net for this, and nobody seems to know. Here is some logging from a *call*: Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Requested indication 3 on channel Zap/5-1 Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '1' Feb 1 17:41:53 DEBUG[6765] pbx.c: Function result...

Hello Everyone, Are these indications of attacks on this system? I specifically have port 22 disabled at all times and only port forward it to server when I access SSH for a minute or so. Shouldn't UNKNOWN be an actual IP address? */var/log/secure:* May 14 00:35:39 pbx sshd[9011]: Did not receive identification string from UNKNOWN May 14 00:36:09 pbx sshd[9040]: Did not receive identification string from UNKNOWN May 14 00:36:39 pbx sshd[9075]: Did not receive identification string from UNKNOWN May 14 00:37:10 pbx sshd[9102]: Did not receive identification string...

Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call e...

We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in the following manner: Current Setup: Telco-> T1->PBX Desired Setup: Telco-> T1-> Asterisk-> T1-> PBX. I am first trying to setup the Asterisk -> T1->PBX part without disturbing existing setup so I can get asterisk to forward c...

...] channel.c: Avoiding initial deadlock for 'SIP/5002-09195d70' Apr 4 17:49:14 DEBUG[6271] app_macro.c: Executed application: Set Apr 4 17:49:14 DEBUG[6271] app_macro.c: Executed application: Set Apr 4 17:49:14 DEBUG[6271] app_macro.c: Executed application: Set Apr 4 17:49:14 DEBUG[6271] pbx.c: Expression result is '1' Apr 4 17:49:14 DEBUG[6271] app_macro.c: Executed application: GotoIf Apr 4 17:49:14 DEBUG[6271] pbx.c: Expression result is '0' Apr 4 17:49:14 DEBUG[6271] pbx.c: Not taking any branch Apr 4 17:49:14 DEBUG[6271] app_macro.c: Executed application: GotoI...

I have a serious problem that repeats very often, after 30 - 50 calls and I can only solve it by stopped and restarting * :-( After a while, * seems to loose track of something. When an ISDN call from PBX needs to go to the Telco, I get 'Ring requested on unconfigured channel 255/255 span 2' It's always channel 255/255 ??, the 'span' number is random Setup: * between Local Telco (2 BRI) and Samsumg PBX (2 BRI). I have also a TDM400P for some Analog lines and have some Sip phon...

Hi, after quite some time doing nothing with asterisk I downloaded the current cvs version. Building this on a SuSE 8.2 System with gcc 3.3 i ran into an unpleasant snag: pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_hangup': pbx.c:4017: internal...

...aching small log snippet from asterisk logs. Thanks Shantanu -------------- next part -------------- [2013-09-25 13:57:33] VERBOSE[1380] netsock2.c: == Using SIP RTP TOS bits 184 [2013-09-25 13:57:33] VERBOSE[1380] netsock2.c: == Using SIP RTP CoS mark 5 [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [09999999999 at from-internal:1] Macro("SIP/1002-00000292", "user-callerid,LIMIT,") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [s at macro-user-callerid:1] Set("SIP/1002-00000292", "AMPUSER=1002") in new stack...

Hello, We have two asterisk PBXs connected. PBX 1 has SIP trunks connected to our provider. PBX 2 is a remote PBX and SIP Trunk connected to PBX 1. We are able to dial extensions either way and PBX 2 is able to dial out using PBX 1 SIP trunks connected to our provider. We would like to use a separated Caller-ID for PB...

...ort (Bri/Pri)) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 No lower Id port:1 init_stack: No such file or directory " Contents of the /var/log/messages for all above commands: Jun 5 20:25:20 pbx kernel: Zapata Telephony Interface Registered on major 196 Jun 5 20:25:25 pbx kernel: Registered tone zone 0 (United States / North America) Jun 5 20:25:48 pbx kernel: Modular ISDN Stack core $Revision: 1.25 $ Jun 5 20:25:53 pbx kernel: ISDN L1 driver version 1.11 Jun 5 20:25:56 pbx kernel: ISD...

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://ww...

...have VBR and VAD (silence suppression) turned off on the soft phone. Here is my SIP debug output of a call from my softphone to voicemail (ext 232 does not answer). Can anyone explain the cutoff? Thanks ------------------------------------------------------------------------ ------------ pbx*CLI> sip debug SIP Debugging Enabled pbx*CLI> Sip read: INVITE sip:232@pbx.ocg.ca SIP/2.0 To: <sip:232@pbx.ocg.ca> From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660 Via: SIP/2.0/UDP 172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport Call-ID: 113d5508a72b5176 C...

...ading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have several analogue extensions spare, but no capacity to add fancier connectors to link to an asterisk system (as most of the PBX linking guides detail). All our phones are bog standard analogue ones. We'd like to use an asterisk...