similar to: Asterisk reject SIP INTITE from different source ports

Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/6114bf1d/attachment.htm

Hi, I am doing failover routing based on 2 dial commands. First route sends back 4xx response and I don't want it to try 2nd route when it is 4xx response. Can we do failover routing based on SIP 5xx response only ? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

----- Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)} Asterisk 1.8 also comes with a 'use_q850_reason' configuration option = for generating and parsing, if available:=20 ----- That will give you what you want if you consider upgrading to v1.8. =09 -----Original Message----- From: asterisk-users-bounces at

Hi, I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi status is not showing alarms but channels are not coming up. It is not showing any channels when i run 'dahdi show channels'. Could anyone help pls. Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have installed R on my computer with windows 7 . I also installed reshape software, but I am not being able to work with melt cast commands . I have chjecked the commands.It is not working. Thankyou, Deepika [[alternative HTML version deleted]]

Hi, My requirement is something like as given below, a.c => a.obj contains a1() and a2() function b.c => b.obj contains b1() and b2() function main.c => main.obj call to a1, a2, b1, b2 Now, I want to move a1(), a2() from a.obj to b2.obj and on top of function b1() When I call b1() from main, it should call first a1, a2 and then function definition of b1 Can you please give me some

Hi, Yes I am looking for IR pass that will do insert call of functions that defined in another file. Links/suggestions that guide me to start for adding IR pass will help me so much. Regards, Deepika On Wed, Feb 10, 2016 at 1:03 PM, mats petersson <mats at planetcatfish.com> wrote: > So how do you know what you want to modify (conceptually)? > > Have you got a IR pass that you

It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100615/e3f6ce8e/attachment.htm

Hi, I want to call/add some functions(that defined in another file) on top of some functions, and reflect the same changes in object file. No, I am not looking for contractor. Thanks, Deepika On Tue, Feb 9, 2016 at 7:04 PM, mats petersson <mats at planetcatfish.com> wrote: > What is the condition for adding this code? > > What have you tried so far? [Or are you looking for a

Hi, I want to edit LLVM generated IR file, like as given below, Original LLVM IR file, @.str2 = private unnamed_addr constant [17 x i8] c"\0AI am in one_11\0A\00", align 1 ; Function Attrs: nounwind define i32 @one_1(i32 %ivar1, i32 %ivar2) #0 { entry: %ivar1.addr = alloca i32, align 4 %ivar2.addr = alloca i32, align 4 %isum = alloca i32, align 4 store i32 %ivar1, i32*

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones

Hi, Asterisk is making a call to a peer. In 200 ok, peer is sending its application server ip in contact field, so asterisk sends ACK to that IP. RTP starts flowing between endpoints and peer plays an IVR and asks for destination number. After entering destination number peer's application server sends INVITE again with different media IP and asterisk accepts with 200 ok. RTP from peer

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Hi All, I am having an ror application with ruby1.8.7 and rails2.3.5, the performance of my application is not good enough. Is their any plugin or gem available to improve the performance. Also I have already optimized some of my code and db queries by optimizing the mysql query and by adding indexes, but those are not gave drastic change in the performance. regards, Loganathan -- You received

Hello, I am running a Samba 4 DC, recently upgraded to the latest version and I have just installed a member server to run as a File Server (Samba 4.1.9). While it seems to be working properly, we are getting a lot of zombie nmbd processes on the member server, running the command *pidof nmbd* results in: *[root at BHFS01 etc]# pidof nmbd* *12861 12644 12404 12236 12071 11885 11720 11553 11388

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This seems to be an old problem, but I haven't found much about it. SuSE mentions "hwprobe=-pci" but, as far as I know, CentOS doesn't use it. This does stop the machine from booting (mounting /) quite a few times. Logs: SCSI subsystem initialized sym0: <895> rev 0x2 at pci 0000:00:13.0 irq 177 sym0: Symbios NVRAM, ID 7,