similar to: Restricting transfers between SIP phones

We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed a change in the behaviour of the queues--a change that we cannot live with. We've used AddQueueMember/RemoveQueueMember to manage logging into and out of our queues for over a year now with Asterisk 1.2, and in that version the queue members were sorted in such a way that the person who had been logged in the longest

Hi, I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues). You can find a lot of info and old firmware versions at this

Hi, I`m using several GXP2020 phones with newest Firmware 1.1.4.18. Asterisk Version: 1.4.11. It happens several times that users complain that the caller cannot hear the transmitted voice from the phones.... Also now it happens quite often that callers on hold beeing dropped. Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name (only IPS configured). I configured

Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest "official" release you can expect a few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped calls). So you have to install the BETAS whether you want or not...

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

https://bugzilla.samba.org/show_bug.cgi?id=5299 Summary: 2.6.9 client cannot receive files from 3.0.0 server Product: rsync Version: 3.0.0 Platform: x86 OS/Version: Windows XP Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy:

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

I have asterisk up and running now, and installed XLITE on 2 PC's. Both machines (mistakenly) registered as the same user / extension. Strangely, asterisks allows this and both phones can make calls! But, only the first one to register can receive calls at the extensions. 1. Is this normal behavior? (Why allow 2 phones on same extension) 2. Why is asterisk not showing the second phone when

On Mon, Jun 26, 2023 at 4:35 PM TTT <lists at telium.io> wrote: > I think that’s getting me close. I’m trying to get (or recreate) the FROM > and TO lines of the header, from a system running PJSIP. I think if I use > CHANNEL to get local_uri and local_tag I can recreate a FROM line like: > > *FROM=<URI>;tag=TAG* > > > > And if I use CHANNEL to get

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> I disagree that it makes it worthless for a large number of JC> users. It's only within the last few days that a few people have run JC> into this particular issue where they have a public IP address that is JC> changing a lot and PJSIP does not support changing it without a JC> restart.

These is my scenario. Asterisk 1.4.16 Zaptel 1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE. Do you know if i can make

I call the firm from my portable at home (zoiper softphone). I have internal extension 60, and I call the internal SIP-client 10 at the firm via an IAX-connection over internet. My colleague at phone 10 answers my call. I ask him to transfer me with my colleague at extension 50. He then presses "transfer" on the grandstream GXP2020 (I get music) and dials the number 50. Phone 50

Hey list ! I'm getting the feedback of a customer that a conversation is like half duplex : when he talks, the other end of the call is no longer heard. What could be the cause of these drop-outs ? A call that is coming in from the PSTN is routed through an IVR-system to the correct internal SIP-phone (Grandstream GXP2020). Where do I start searching for this problem ? -------------- next

Hi, I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but with 2 extensions BLF status does not work correctly. have someone ever tested a Sip Phone with more then 60 BLF without problems? Can someone suggest me model and brand? Thanks, bye. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

Hi, I am in a weird situation where a variable seemed ignored, but not always. That variable is __TRANSFER_CONTEXT. Basically, I have a phone registered with asterisk. It's context is "internal". Outgoing calls go through that context (all good). When I get an incoming call which I want transferred, I don't want it to go through the context "internal" but