similar to: Remote Party ID issue

Hello, Did anyone manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

On Tue, Mar 13, 2018 at 2:42 PM, Ondrej Valousek < Ondrej.Valousek at s3group.com> wrote: > Yes, I have had this in place already (well except of the negative cache, > but enabling that did not make much effect). > > To me, this is no surprise ? nothing can match nfs performance for small > files for obvious reasons: > Could you give profile info of the run you did with

>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing

Yes, I have had this in place already (well except of the negative cache, but enabling that did not make much effect). To me, this is no surprise ? nothing can match nfs performance for small files for obvious reasons: 1. Single server, does not have to deal with distributed locks 2. Afaik, gluster does not support read/write delegations the same way NFS does. 3. Glusterfs is

Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice ----------------> Asterisk ------------------->Bob ------- INVITE

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

We can't stick to single server because the law. Redundancy is a legal requirement for our business. I'm sort of giving up on gluster though. It would seem a pretty stupid content addressable storage would suit our needs better. On 13 March 2018 at 10:12, Ondrej Valousek <Ondrej.Valousek at s3group.com> wrote: > Yes, I have had this in place already (well except of the negative

Well, it might be close to the _synchronous_ nfs, but it is still well behind of the asynchronous nfs performance. Simple script (bit extreme I know, but helps to draw the picture): #!/bin/csh set HOSTNAME=`/bin/hostname` set j=1 while ($j <= 7000) echo ahoj > test.$HOSTNAME.$j @ j++ end rm -rf test.$HOSTNAME.* Takes 9 seconds to execute on the NFS share, but 90 seconds on

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

On Tue, Mar 13, 2018 at 1:37 PM, Ondrej Valousek < Ondrej.Valousek at s3group.com> wrote: > Well, it might be close to the _*synchronous*_ nfs, but it is still well > behind of the asynchronous nfs performance. > > Simple script (bit extreme I know, but helps to draw the picture): > > > > #!/bin/csh > > > > set HOSTNAME=`/bin/hostname` > > set j=1

When receiving inbound calls from a Vonage Softphone extension, I'm unable to view/maniupulate calledid data. but it shows up in the CDR records and on called handsets.. any ideas? exten => asda,n,NoOp(callerID is ${CALLERID}) exten => asda,n,NoOp(CallerID is ${CALLERIDNAME}) exten => asda,n,NoOp(CallerID is ${CALLERIDNUM}) -- Executing [asd at pstn-in:2]

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

That seems unlikely. I pre-create the directory layout and then write to directories I know exist. I don't quite understand how any settings at all can reduce performance to 1/5000 of what I get when writing straight to ramdisk though, and especially when running on a single node instead of in a cluster. Has anyone else set this up and managed to get better write performance? On 13 March

Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't send user's names. I also want to have reverse callerid for avaya users. But if there is no answer in dial plan: exten => _XXXX,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})}) ;exten => _XXXX,n,Answer exten => _XXXX,n,Dial(DAHDI/g4/${EXTEN})

Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I