similar to: Secondary redirect failed

? Hello all, we will bridge the two asterisk call, after that i am trying to redirect the call to ivr. here i faced some problem 1)originate the first call sip/1-234 2)originate second call sip/2-245 3)bridge both the call 4)redirct both the call to IVR the call has been hangup..... which cammand i have to use in asterisk manager API Regards, Pandi.P -------------- next part

Hello, Please copy all replies directly to me (my account is having difficulty with receiving lists these days). This is primarily a programming question, but the specifics regard start values for a nonlinear regression (if you have suggestions on alternative ways to obtain start values they are welcome as well). I'm using nls to estimate a nonlinear time series equation of the form:

I am using AMI to call a phone and play a wave file. That works fine to SIP/401. Now I am trying to "redirect" that call that is ringing to another phone (SIP/404). When I do it the other phone rings but the first phone continues to ring also. Then when I answer on SIP/404, I get a ring not the wave file. Action: Setvar Channel: SIP/401-00000004 Variable: SMVOICE_CALLAT

I didn't get any response on that question, so i supose this feature is possible but there isn't an implementation of it. I'm ready to sponsor this feature in the manager interface (i tried the redirect command but it doesn't work) can somebody help me ?? this feature would make it possbile to use drag & drop features ... Kind Regards Michael Devenijn

Hello, I'm using the classic [stdexten-macro] in extensions.conf whereby a call is picked up by voicemail after a certain ringing time. When programming a SIP phone to redirect calls (SIP 302 redirect) to another extension I'd like to avoid that voicemail pickup so that the call goes into the new destination's voicemail (if applicable). How can I detect that a call has been

is it possible to transfer a call while it's ringing ?? SIP/cs001 calls SIP/cs002 The SIP/cs002 user transfers the call to SIP/cs003, where on SIP/cs003 the phone continues to ring ... in one way or another (trough manager API or something else, don't care) ???? i tried redirect with the manager but it doesn't work (or i didn't understand it ) Thank you for any help

Hi, The ami manager call out with an originate through dadhi to a local number (A). If this call is answered, then the ami manager redirect this call to a dial command. This dial command calls through dadhi to another local number (B). Number B answers this call and number A en B are connected. If number B and number A hangs up, there is will be no CDR be written If the dial command is commented

Hello all,   I am using HALF DUPLEX modem for TAPI call.the following message is displayed while i am starting the AsteriskNOTICE[1416] chan_tapi.c: Channel format set to ULAW\' ERROR[1416] win32_tapi.c: TAPI Error: 80000023 (HCALL 0x0) on lineGetID . If i will receive an Inbound call to modem, i will answer that call............and put an wait for infinite time.The

? Hello all, We are setting up a gateway in which the SIP devices will be connected dynamically using the Asterisk system. We use the originate Manager API command from our code to call an IP as (SIP/1@10.20.30.40). The call rings on the phone and goes through the normal (default) context and finally hangs up(WARNING[13833]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context

Hello, I'm running Dovecot 2.0.13 on my Ubuntu10.04. It's working very well but I faced a problem with Redirect and Vacation using Sieve. :-( In fact, after well configuring the system I dont't receive the redirect message, and in the log I found: ######################################################################################## Jun 9 18:49:37 box7 dovecot: lmtp(10734, user at

adding a second, non-redirect action to the sieve rule in order to test, + require ["copy","fileinto"]; # rule:[SIEVETEST] if header :contains "subject" "SIEVETEST" { + fileinto :copy "testing"; redirect "user2 at example2.com"; } on send exec, the 'fileinto' action does work as expected. the 'redirect' fails

Hi Guys, I have a strange problem and I couldn't find any solution, I hope somebody could help me. I'm using postfix+dovecot+sieve combination and I tried to set up redirect. .sieve: require ["fileinto", "regex", "date", "relational", "vacation"]; redirect "xxxx at gmail.com"; keep; But I get some error in the mail.log: sieve:

Hi Andreas, note that with the introduction of ALTREP, as far as I understand, calls as "simple" as DATAPTR can execute arbitrary code (R or native). Even without ALTREP, if you execute user-provided R code via Rf_eval and such on some custom thread, you may end up executing native code of some package, which may assume it is executed only from the R main thread. Could you (1)

When I use the Asterisk manager interface to redirect a call (Action: Redirect) I get an error with the message "Redirect failed, channel not up." This is especially troubling as it looks like this message was added to the code for the rather recent 1.2.x release. A quick google search implies that I'm not the only one experiencing this problem with 1.2.17, but me and

I run dovecot --version 2.3.10.1 (a3d0e1171) postconf mail_version mail_version = 3.5.7 with three valid accounts on the server, user1 at example1.net user2 at example2.net admin at mx.example.com I can send/receive from each -- both to/from external addresses, as well as between one another. E.g., on mail sent to user2 at example2.net -> admin at mx.example.com logs of the

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

We have a situation where we get no early media in this call flow: VoIP origination provider Server1 (our server) Customer server Customer phone with call-forward set Server1 to dial the forward-to number Then there is no early media while the forward-to number is ringing. Our server is Asterisk 1.6 and theirs is 1.8. I tried promiscredir=yes and then the calls fail altogether because rather

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks First of all, excuse my bad english. I'm working in the video streaming team of the next WikiMania, the conference of the Wikimedia fundation, this year in Buenos Aires. We are facing some issues with some specific features, the setup we are thinking of is the classical dvgrab | ffmpeg2theora | oggwd * It all works fine, but we need

Stepan, Andreas gave a lot more thought into what you question in your reply. His question was how you can avoid what you where proposing and have proper threading under safe conditions. Having dealt with this before, I think Andreas' write up is pretty much the most complete analysis I have seen. I'd wait for Luke to chime in as the ultimate authority if he gets to it. The

Sorry about a bit offtopic, but I am looking reliable (not free) secondary dns provider. -- Eero