search for: myasterisk

...23000: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry '0000-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) [Apr 25 10:56:56] WARNING[19013][C-000002cb]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to MyAsterisk-asterisk [MyAsterisk-asterisk]... [Apr 25 10:56:56] WARNING[19013][C-000002cb]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 25 10:57:01] NOTICE[19013][C-000002cb]: res_odbc.c:1527 odbc_obj_connect: Connecting MyAsterisk-asterisk [Apr 25 10:57:01] N...

...ubuntu0.14.04.1-log]Duplicate > > entry '0000-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) > > > > [Apr 25 10:56:56] WARNING[19013][C-000002cb]: res_odbc.c:657 > > ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying > > connection to MyAsterisk-asterisk [MyAsterisk-asterisk]... > > > > [Apr 25 10:56:56] WARNING[19013][C-000002cb]: res_odbc.c:761 > > ast_odbc_sanity_check: Connection is down attempting to reconnect... > > > > [Apr 25 10:57:01] NOTICE[19013][C-000002cb]: res_odbc.c:1527 > > odbc_obj_conne...

...1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry > '0000-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) > > [Apr 25 10:56:56] WARNING[19013][C-000002cb]: res_odbc.c:657 > ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to > MyAsterisk-asterisk [MyAsterisk-asterisk]... > > [Apr 25 10:56:56] WARNING[19013][C-000002cb]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... > > [Apr 25 10:57:01] NOTICE[19013][C-000002cb]: res_odbc.c:1527 odbc_obj_connect: Connecting MyAsterisk-asterisk >...

Okay...here's one for all of you.... 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. C...

...l Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Adrien Laurent > Sent: 02 August 2005 14:56 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] How to create a secret code to use > myasterisk@home server's long distance plan from a public phone > > > Hello everyone, > > > I have an IAX server (asterisk@home) with a FXO card. > I have a trunk connected to a voip provide, asteriskout. > > When I call my server from a public phone, I want to route this c...

I'm not sure how much more simple I can make this but I just cannot seem to get my Asterisk 13 to accept a connection on the manager interface: --- manager.conf --- [general] enabled = yes port = 5038 bindaddr = 127.0.0.1 [myasterisk] secret=a permit=0.0.0.0/0.0.0.0 read = all write = all So, couldn't be any more wide open and simpler to connect yet: # echo -e "Action: Login\r\nUsername: myasterisk\r\nPassword: a\r\n\r\n" | ncat 127.0.0.1 5038 Asterisk Call Manager/2.10.4 Response: Error Message: Authentication...

Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP those calls refer to? The above literature don't say much to help...

On Thu, 28 May 2015 11:15:45 -0500 Scott Griepentrog <sgriepentrog at digium.com> wrote: > The string "5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060" is the unique > identifier for the call in SIP known as the Call-ID. If you have a packet > capture of the port 5060 SIP traffic, that identifier will be in each SIP > message related to the call, which also includes the full from and to > details. That is the problem. Sinc...

The string "5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060" is the unique identifier for the call in SIP known as the Call-ID. If you have a packet capture of the port 5060 SIP traffic, that identifier will be in each SIP message related to the call, which also includes the full from and to details. As an alternative to running a separate packe...

...hG4bK3afa3dd6;rport// //Max-Forwards: 70// //From: <sip:10 at ip-sip-server;user=phone>;tag=as00df4bee// //To: <sip:testacc77003 at ip-sip-server>;tag=9wdraz254n// //Contact: <sip:10 at ip-sip-server:5060>// //Call-ID: 3c267066aeb1-bv3r703hb93x// //CSeq: 109 NOTIFY// //User-Agent: myasterisk// //Subscription-State: active// //Event: dialog// //Content-Type: application/dialog-info+xml// //Content-Length: 202// // //<?xml version="1.0"?>// //<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="7" state="full" entity="sip...

Hi.. Iam new with asterisk PBX, and i have read about asterisk video call.: my question: 1. Is imposible to establish system video call (from Phone with GPRS/3G enabled to Computer Running Softphone like X-Lite) over Asterisk Gateway.. 2. If posible what requirement (Hardware and Software on my Asterisk,PC or My Phone) Thanks Joko Pitoyo -------------- next part -------------- An HTML

...iginal Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 27, 2009 3:59 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] The Mobile devices are not able to register on myasterisk Dear All; I am facing a problem that all the mobile devices that support SIP and are able to register with a lot of providers, they are not able to register on my asterisk. What could be the reason? Any specific thing I have to do? The used port is UDP 5060 Actually, any SIP Phone can register...

...hat Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes ; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srvlookup=yes And here's an example of a ws client in my realtime peer table: id: 4 name: 660...

Hello.. How is this possible?? I have 65 active calls .. but making new calls and registering isn't possible anymore the CLI command restart now didn't even work .. had to kill the process before it worked again... myasterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) 65 active call(s) Jun 27 16:22:06 WARNING[20313]: channel.c:531 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/mistered-ddb7', 10 retries! -------------- next part -----...

...Via: SIP/2.0/UDP xx.xx.xx.98:5060;branch=z9hG4bK168884d7;received=xx.xx.xx.98;rport=5060 From: "5006" <sip:5006 at xx.xx.xx.98>;tag=as50c98b4c To: <sip:0419 at xx.xx.xx.238>;tag=as3c6e57b0 Call-ID: 6d1039bb22716c6e6dec69fb3e78a8d7 at xx.xx.xx.98:5060 CSeq: 102 INVITE Server: myasterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 How can I make this work ? Thanks. Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asteri...

...ime => 60 I've done that since a customer asked me a such function. Feel free to try it. Disclaimer : I haven't tested it heavily... just seems to work ;) Matteo. -------------- next part -------------- --- asterisk/res/res_parking.c 2003-07-02 16:06:12.000000000 +0200 +++ myasterisk/res/res_parking.c 2003-07-03 23:28:52.000000000 +0200 @@ -49,6 +49,7 @@ /* Extension you type to park the call */ static char parking_ext[AST_MAX_EXTENSION] = "700"; +static char parking_pick[AST_MAX_EXTENSION] = "750"; static char pickup_ext[AST_MAX_EXTENSION] = "*8...

This is coming from an * noob. :) I've got two customers, they both are replacing their phone systems with VOIP, and we need to retain both their existing dialplans. One has 5 extensions starting at 100, and the other has 10 extensions, starting at 100. Is there a way to have the same extension number twice in the same asterisk system ? They will have different incoming DIDs of course.

...erisk on FreeBSD + Passive ISDN BRI (Cian Hughes) 7. Re: How can I get a recording from a CD to my asterisk digital assistant (Alberto Sagredo) 8. Re: PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3 (John Novack) 9. Re: How can I get a recording from a CD to myasterisk digital assistant (Davi-Ann) 10. Re: RE: SPA 3000 - UK Replacement (Wayne) 11. Re: Sipura SP3000 question (Wayne) 12. RE: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?" (Steven Totaro) 13. RE: Pinouts f...

...nal list > ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) > ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule > ;ignoreregexpire=yes ; Enabling this setting has two functions: > ; domain=myasterisk.dom > ; domain=customer.com,customer-context > ; autodomain=yes > ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to > > #include sip-vicidial.conf > > ; register SIP account on remote machine if using SIP trunks > ; register => testSIPtrunk:test at 10.10.1...

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk receives only the "foo" part. If someone calls john@doe.com, it receives "john" as the extension. Now the main question is: how do I know which SIP address the call originally wen...