similar to: Understanding of SIP Info Messages

2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com> > Kevin P. Fleming wrote: > > Olivier wrote: > > > > > >> 2. R Hook-flash key is now available to transfer calls. > >> In s450IP web management server, its defaults settings are : > >> Application-type: dtmf-relay > >> Application-signal: 16 > >> >

Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb

Sorry if this was posted yesterday, I was having issues with being auto-unsubscribed because of my spam filter. Not sure if my post made it through. Hi everyone, I'm wondering if I'm missing something obvious here, or if Asterisk just doesn't support what I'm trying to do. It seems like it should be simple, but appearances can be deceiving. I've got an Asterisk box

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb

Hi all, I`m searching for a special solution to send text messages inside Sip info packets, that are normally used for dtmf signalization. So far I?m able to exchange sip Info messages between two softphones which are connected directly together (only over a Switch). By connecting both Softphones on the asterisk pbx, registration is ok and the voice interconnection is also fine. During the call,

Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb

Hi List I'm trying to update to the lastest kernel but I have a dirver that is not inculded in the distrubution, and I had to use the driver disk when installing centos 4.4 in the first place, The driver megasr .ko works fine with the installed kernel but I cannot find on for the updated kernel, any adive would be appreciated. without the updated driver there is a kernel panic on boot due to

Hello List, We are facing a problem in broadcasting DTMF from MeetMe. Our client is sending DTMF with duration=160 (in SIP INFO) to asterisk but asterisk is changing this header to different values like 162, 175 etc while broadcasting to all the participants. Is it possible to restrict asterisk from changing this header value or this is a common behavior of all the PBXs. Regards, Rajib

Hi All BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with

Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways,

Hi I have a problem with a new DECT phone I have bought The key pad works like a Mobile phone where you dial first then pick up the line, but it seems to dail too fast or spuriously, ie 012826736464 show on thew Asterisk console as 0012282677, could any one offer advice how to fix? Also when doing a ZAP bridge to this phone from an outside line the call is very echoy, but not an internal

Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at "Starting udev" if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb

Robb, I wrote up a small tutorial on setting up the standard tftp server for linux check it out on my site. http://asterisk.titaniumsoft.net/ Mitchel -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Robert Boardman Sent: Thursday, May 13, 2004 2:44 PM To: asterisk-users@lists.digium.com Subject:

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk