similar to: Asterisk SIP UUI Protocol

Dear all, How do I set the UUI informations for outgoing calls and read the UUI information for incoming call in asterisk? Thanks in advance.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/ffcceeee/attachment.htm

Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know

Hi there, Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card). Basically, is it possible to send User to User Information using the D-channel, while making a call?

Hello List, I need to connect with an Avaya PBX (this part is done), and i would like to get and send back User-to-User Information (UUI) with the call. The UUI need because I need to identify the call based on something witch is available on Asterisk and Avaya too. It is possible, or have a better solution? Anybody did it before? Thanks for the help! Zsotya

OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help.

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

Hi, After reading the source code of openssh and man page of sftp. In sftp.h it define 27 /* 28 * draft-ietf-secsh-filexfer-01.txt 29 */ 30 31 /* version */ 32 #define SSH2_FILEXFER_VERSION 3 and In the end of the man page, T. Ylonen and S. Lehtinen, SSH File Transfer Protocol, draft-ietf-secsh-filexfer-00.txt, January 2001, work in progress material. In wikipedia of "SSH File

OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Its really weird working with OpenSuse. I am not sure how others are using > with OpenSuse. Through Yast also I tried to install Asterisk package, it > didn't find. > > Now I am clueless to work with OpenSuse. > >

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=

Hi, If the special handling in the meg2reg pass is to look for lifetime intrinsics, shouldn't it cast to <IntrisicInst> and then use getInstrinsicID to check for lifetime_start and lifetime_end ? The thing that I don't understand is the following piece of code, which finds all the users and cast it to <Instruction> then eraseFromParent(). How can this guarantee that it only

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

What happens when we increase the queue frame size in channels.c if ((queued_frames + new_frames > 128 || queued_voice_frames + new_voice_frames > 96)) { Be default it is 128 and 96 if i increase it to 256 and 192 what will happen? will it impact to default behavior? Regards, Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Timo, I joined the lemonade list for the new IMAP sending protocol and I mage the following suggestion about using IMAP as a front end for SMTP. The idea would be that a client could authenticate to IMAP, then ask for a tunnel to the SMTP server. - Here's my message. I thought I'd get people thinking. --------------------- OK thanks. I would clarify what I'm proposing. I looked

Hello CentOS Community Members, What is RH's be-all end-all justification for staying with an ancient code base for such important programs as BIND et al? A similar problem (to BZ561299) was first reported five (5) years ago on the isc.org mailing list. Is there any support among the CentOS community for a REPO of current vintage for such important functions as BIND et al? That question is

Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

I spent some time today implementing support for loading U2F keys into the SSH agent from my AsyncSSH library. I got it working, but along the way I ran into a few issues I wanted to report: First, it looks like the value of SSH_AGENT_CONSTRAIN_EXTENSION has changed from the value 3 defined at https://tools.ietf.org/html/draft-miller-ssh-agent-02

Hi, Asterisk 1.4.17 Sangoma a102DE I'm having some issues sending CallerID Name to a Dialogic based phone app. According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see that it is sending the CallerID Name "Mike - Budgetone - reachme.com" to the Dialogic card, but it isn't regestering on the Dialogic based system. I can receive CallerID Names from our Paetec,