similar to: early media (video)

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3.

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

Hello, I am looking for a consultant that know asterisk in and out including how to troubleshoot sip and rtp. I have a device that this acting very strange and I need to prove it's the device code and not an issue with my setup. Very simple setup, all local no nat... Grandstream video phone and a AIphone IX-MX7 door station. PJSIP ... doorstation to grandstream 3370 works perfectly. Early

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

Hello, My target system is : PSTN <---> Sip Provider <---IP/ADSL---> Router with fw/NAT <--- SIP/IP/Eth --> Asterisk <--- SIP/IP/Eth --> SIP Phones Asterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I don't have any problem (yet) with either casual incoming or outgoing calls. To work around a possible No Audio when an incoming

Am 2015-09-24 um 20:32 schrieb Marc Muehlfeld: > Hello Matthias, > > Am 23.09.2015 um 10:45 schrieb Matthias Leopold: >> this seems to be a mat{,t}hias discussion ;-) > > oh, sorry. :-) > > > > >> Marc, you said i can have windows ACLs on a standalone server. How do i >> accomplish the >>

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

Can you better this String acronym method? def acronym name letters=[] name.each_char {|char| letters<<char if char[0]>=65 and char[0]<=90} acronym = letters.join " " end chris -- Posted via http://www.ruby-forum.com/.

Hi All, I want to refresh a partial onClick, onClick i am making ajax call and getting the data but i am unable to refresh the partial. here is the code IN views: home.html.erb $(document).ready(function() { var currentCellText; $(".inline").click(function() { currentCellText = $(this).text(); $.ajax({ type: ''GET'', dataType: "json",

I have problems running Monkey Island 3 here. Installing it works fine, but the problems comes when trying to run it. When pressing "Play Game" (I think that's what the button says), the program just wants to install DirectX 5.0. I guess that trying to install it is NOT an option, when you're using wine. Second, installing Monkey Island 4 is not successful, because you have this

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

Hello, I have question regarding the changes that are made in the sip protocol in Asterisk - the option progressinband. When this option is set to yes in asterisk version 1.4.21.1 - the call flow is: sip.conf: progressinband=yes Device Asterisk -----------INVITE SDP---------> <---------100 Trying------------ <-----183 Session Prgoress-- After version 1.4.2X+ (tested

Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg "30% increase") that would be great, rather than just "lots". Also, are there any

I maintain a directory structure containing dirs and files that I regularly push to ~50 hosts, which are divided into 3 groups that have slightly different needs (minor mods in a couple of files). So ideally I would have 4 directories: /path/to/sync/common/ <- common files /path/to/sync/group1/ <- group1 specific only /path/to/sync/group2/ <- group2 specific only

Whoops, email client auto-filled dev previously. Let's try this again. Michael Maier wrote: <snip> > Ok - but this doesn't seem to answer my main question: > > Why must > > progressinband=never > > be applied especially if asterisk uses it by default? The big difference > between w/ and w/o it is: The default in 13 is "no" which still

Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper