similar to: Re: 482 Loop Detected on sip calls

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

I'm having some difficulty with chan_local dial requests. It seems that when a chan_local call is picked up, that the native bridge "pops" the environment back to the settings of the original call. This is unexpected and leads to very frustrating results. My example below is a very distilled sample of a much more complex dialplan problem I'm having with chan_local, but it

Hi, I wish to use the manager API to make an outbound call to a sip url,subsequently play a prompt and hangup.Any hints on how to acheive this/feasability will be much appreciated. Regards, Ajit

Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP -> SIP SIP -> ZAP Not working (no jitter buffers created): SIP -> chan_local (with /nj) -> ZAP SIP -> chan_local (with /j) -> ZAP SIP -> chan_local (with no flags) -> ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make

Hi, I am trying to use the R package 'spatstat' for generating spatial poisson point process graphs. I can create a point pattern using the following commands: pp <- rpoispp(.01, win=owin(c(0,100),c(0,100))) and also view the resulting graph by: plot(pp) But how can I export the generated point pattern to an external file so that I could use it as input for some network

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

Hi, am building the latest version of Asterisk (1.4.13) on a self-build Linux host (based on LFS-6.3). Version 1.4.12 built installed and worked fine. Last night I upgraded the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1 against it. That seemed to build and install O.K. too. I dl'd the 1.4.13 source tarball and tried to build that: ./configure ran O.k. 'make

Hello R-world, I am facing a peculiar problem and hope someone out there can comment on it. In goodness-of-fit tests for evaluation of distributions, there are three well-known methods: 1. Chi-square 2. Anderson-Darling 3. Kolmogorov-Sminrov I am trying to use the second test. Many researchers have reported results using this test. I wrote programs in C and now in R to do this. I run into

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 10000 - 30000 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see

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Greetings: I have formed a group on the LinkedIn.com professional networking site. There are jobs listings and discussions there. Not exactly code discussions but some professional topics including favorite GUIs and trainings. http://www.linkedin.com/groups?about=&gid=77616&trk=anet_ug_grppro Regards, Ajit -- _________________________________ Ajit Gemunu de Silva Oakland CA 94619

I have an app which handles a Mitel's command port to change the MWI lights. It detects dial tone, plays some DTMF digits, listens for dialtone-or-busy, does a manager event on what it finds, and returns. Since the Mitel command port does not give answer supervision (looks like it's ringing), and since I run this app via a AMI "originate" command, I set up an extension in

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Hi, I want to use USDT in my java applications. Please tell me how to achieve this. I am comfortable with SDT in C application. I am using Solaris Sparc10 with Generic_120011-14 version. Do I need to update my machine for this? Thanks, Ajit -- This message posted from opensolaris.org

Hello asterisk users, Compiled asterisk-11.13.0 on openSUSE 13.1, however Channel driver chan_sip is XXX in menuselect --- it depends on: chan_local(M), res_crypto(M), res_http_websocket(M) chan_local is [*] chan_local in menuselect, res_crypto is in Resource Modules, Depends on: openssl(E) --- I don't know what (E) means ??? res_http_websocket is [*] res_http_websocket in

Hi all, If any perticular OS does not have 'struct cmsghdr' in sys/socket.h and also it does not have access rights in 'struct msghdr', then how the compilation should be done ? Does anybody had similar problem earlier ? Regards, Ajit

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? I Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So, I've configured some keys to transfer the calls like this: [featuremap] blindxfer => #2 ; Blind transfer (default is #) disconnect => *0 ; Disconnect (default is *) ;automon => *1

Hi, The following controller method taking me into infinite loop. Once the update action completes I want to reload the ''index'' page. May I know why it is going into infinite loop? def update Device.find_by_id( params[:device_id] ).driver = ( params[:driver_id] == 0 ) ? nil : Driver.find_by_id( params[:driver_id] ) redirect_to :action => :index, :tab =>