similar to: Early Media in 180 Ringing

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route:

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

Hi All, I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has extension number 600. Now what I want to accomplish is the following: - If a mobile-number is chosen by a user, asterisk needs to call the ATA (600), wait for a few seconds, and then send the mobile-phonenumber. Or, if it's possible, define the

I'm trying to implement a shared asterisk server for multiple (different) companies. Here's what I've done so far: - I've installed multiple asterisk instances on one server (via vserver). Each * is for one customer, and has it's own extensions (like 100, 101, 102, etc.) Note that the same extension can exist on other * instances - The SIP Clients register themselves with * -

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able

For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore

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Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

Tim Legge has passed the following message on to me: hpa wrote: > Quinn wrote: > > Now it seems I have found a bug in PXELINUX with regards to etherboot. > > Up until 2.07 pxelinux was responsive to the keyboard. From 2.08 and > > up the keyboard is locked at the boot prompt when pxelinux is loaded > > via etherboot 5.3.14. > > That would not be a PXELINUX bug,

On 10/06/2019 08:51, Tom?? Havl?n wrote: > Hello > my smb.conf + working and no working ACL share folders > > [global] > netbios name = FENIX > realm = PFCZ.INTRA > server role = active directory domain controller > workgroup = PFCZ > idmap_ldb:use rfc2307 = yes > dns forwarder = 10.254.254.1 > > unix extensions = no > wide links = yes > follow symlinks =

Hello I'm working on my RIS server; I got a problem when I use pxelinux.0 to load setupldr.exe from windows XP 32bits (both home and corporate) Here is how I configure a working server, then how to modify it to reproduce the bug: ========================= I suppose you got -a dhcp server configured to load pxelinux.0 to client computer. -a working tftp server (mine is tftpd-hpa, with a rule

Rowland, You're absolutely right - with '128' both DHCP servers listen and *may* reply, therefore besides failover provide load balancing. Even if both servers respond with *Offers* to the client, the client will select one and issue *one Request* to the selected server. E.g., below is a log records from my failover DHCP (192.168.0.2) server: May 17 00:09:37 s-dhcpd-2 dhcpd[1578]:

Hi All, Looking for some guidance on debugging and also reporting a bug. First off I can't seem to find any information on reproducing the ubuntu 16.04 build with debug symbols, I do have a core file I'd like to introspect to potentially provide a patch. Secondly the bug report itself: I am receiving a segfault signal 11 when one or more machines share a *Address* specific IP. For

On 21/05/2020 21:04, Matt Savin wrote: > I think you can continue to use '128' (good for load balancing, > otherwise 255 is fine). I had the same issue, following dhcpd.conf > entries fixed it (the last entry is important): > > one-lease-per-client? true; > deny ? ? ? ? ? ? ? ? ? ? ? ?? duplicates; > ignore-client-uids ? ?? true; > > And it's not a bug,

I had similiar issue on samba 4.8 domain member (new files with wrong permissions), when I realised that You need to list all modules that You wish to use in "vfs objects" every time , there is no inheritance from global -> shares that is if You have e.g. [global] ... vfs object = acl_xattr .. [some share] ... vfs object = recycle .. On samba DC "acl_xattr" is