similar to: Does IAX2 support call completion or callback ?

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

Hello, I am attempting to write a script to launch a callback based on a dial-in service. I have created this call file: --------------------------- channel: IAX2/user@voipjet/011_valid_number maxretries: 3 retrytime: 5 waittime: 5 context: dialtone extension: 912125551212 priority: 1 --------------------------- Where I first attempt to dial the callback user (channel) and then connect the

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

On Sun, Jul 16, 2023 at 04:39:18PM +0000, Tage Johansson wrote: > > @@ -194,7 +198,10 @@ calls. The cookie is unique (per libnbd handle) and E<ge> 1. > > > > You may register a function which is called when the command > > completes, see L</Completion callbacks> below. In this case we have > > -specified a null completion callback. > >

On 9/5/20 7:47 AM, Eric Blake wrote: > I noticed while reading the code that we have a documentation hole that > may cause memory leaks for clients that are unaware, in relation to > completion callbacks. > > The situation arises as follows: for all commands with a completion > callback, I checked that the code has clean semantics: either > nbd_aio_FOO() returns -1 and we

We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional [May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211) [May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded

On 7/14/2023 4:13 PM, Eric Blake wrote: > On Fri, Jul 14, 2023 at 09:13:42AM +0200, Laszlo Ersek wrote: >> On 7/13/23 21:29, Eric Blake wrote: >>> The documentation has claimed since commit 6f4dcdab that any >>> completion callback will be called exactly once; but this is not >>> consistent with the code: if nbd_aio_* itself returns an error, then >>>

Hi, I'm trying to set up a call-back system using auto-dialout files. I want the call to be terminated when a specific timeout (defined in the .call file) is detected. Both parties should then be hangup. The problem is that the timeout is never detected... How to solve this? Thank you, Pierre .call file ---------- Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111 Callerid: 111111111

Squash this into opt_go? into opt_info? Standalone? Testing: why does python not throw an exception: $ ./run nbdsh Welcome to nbdsh, the shell for interacting with Network Block Device (NBD) servers. The ‘nbd’ module has already been imported and there is an open NBD handle called ‘h’. h.connect_tcp ("remote", "10809") # Connect to a remote server. h.get_size ()

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

We are using Asterisk 1.0.7. We have this scenario: IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user. The dial command is simple, looks like this: exten => 300,1,Dial(SIP/300) Extension 300 is a legacy PBX device operated by touchtones. The user (coming in over IAX2) is trying to drive this PBX using touchtones. But the trouble is, by the time the touchtones

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

Hello, Just in case someone hasn't upgraded yet, and is using IAX2. -------- Original Message -------- Subject: AST-2009-006: IAX2 Call Number Resource Exhaustion Date: Thu, 03 Sep 2009 17:47:35 -0500 From: Asterisk Security Team <security at asterisk.org> To: bugtraq at securityfocus.com Asterisk Project Security Advisory - AST-2009-006

I noticed while reading the code that we have a documentation hole that may cause memory leaks for clients that are unaware, in relation to completion callbacks. The situation arises as follows: for all commands with a completion callback, I checked that the code has clean semantics: either nbd_aio_FOO() returns -1 and we never call the callback cleanup, or nbd_aio_FOO() returns a cookie and

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

Hello, In my asterisk box, i have a zaptel card connected to my analogic pstn line. I'm using a IAX2 client to call outside : IAX2 client <--> Asterisk <--> Zaptel card <----> France telecom line When checking cdr logs file, i always have an "ANSWER" on call status when call on this trunk, even if the final destination does not answer. Is "ANSWER" the

Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols