similar to: SIP call hangs up after 20 seconds

Hi- I'm at my wit's end trying to get a Grandstream BT-101 POS to register on my * server. Asterisk version is 1.2.1. GS Firmware is rev 1.0.6.7. Basically, I've setup the phone following the instructions at voip-info.org, and it registers for about 10 seconds, then after receiving the SIP NOTIFY from the * server, goes into "flashing display" mode, which indicates some

Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a "graceful restart"). Besides, would a "restart gracefully" actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via ->QueueAdd("4050", "Local/4053 at from-internal/n", 1) (not via agents.conf). SIP extension 4053 rings, answers and then decides to blind-transfer to ZAP/DAHDI extension 3666. The "show queue" command still displays 4053 as "In use". However, if 3210 calls 4050

Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so Asterisk server 1 (192.168.250.111) sip.conf contains: [interboxsip] type=peer host=192.168.250.112

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

FWIW, now that privsep is mandatory I have no objection to including signal support in sshd. On Wed, 25 Jul 2018, Yonathan Bleyfuesz wrote: > Hi all, > > I would like to propose some ideas to revivify this subject. > > -First, we could add support on the client to send signal thanks to the escape characters. > (code :

I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten => **101,1,NoOp(pickup extension) exten => **101,n,Pickup(101) exten => **101,n,NoOp(pickup group) exten =>

Without call-limit defined, when a sip extension calls another sip extension then "show hints" shows that both are InUse (as expected). When one of them hangs up, both hints status become "Idle" (as expected). With call-limit=1 for each SIP extension: the caller is always Idle while the callee is InUse. Is this behavior normal? Doesn't sound right because if, during the

On 21/08/2019 17:31, Robert A Wooldridge via samba wrote: > On 08/21/2019 02:02 AM, L.P.H. van Belle via samba wrote: >> Try this command: >> samba-tool domain join edm-inc.com DC \ >> ????--server=server.fqdn.here \??? # << AD-DC server with FSMO roles >> ????--realm=EDM-INC.COM >> ????--dns-backend=SAMBA_INTERNAL \ # if your running with bind9, >>

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

On 2019-04-05 08:57, David B?rgin via dovecot wrote: > Andr?, are you quite sure you have it working? > > In this thread someone from Open-Xchange stated that no, Dovecot > doesn?t > have SMTPUTF8 support implemented, and the same response was given by > another Dovecot developer last September (it ?is being considered? was > the answer then, see >

On 08/21/2019 02:02 AM, L.P.H. van Belle via samba wrote: > Try this command: > samba-tool domain join edm-inc.com DC \ > --server=server.fqdn.here \ # << AD-DC server with FSMO roles > --realm=EDM-INC.COM > --dns-backend=SAMBA_INTERNAL \ # if your running with bind9, --dns-backend=BIND9_DLZ > --option='idmap_ldb:use rfc2307 = yes' \ >

Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register:

I have a mixed PBX system with both Asterisk 1.4.21 and 1.2.27 (moving to 1.2.28). For now I need to keep a few boxes in 1.2 and not migrate them all to 1.4. However, I would like to have func_odbc and res_odbc on all servers. On 1.4.21, native func_odbc seems to work fine. On 1.2.27, the func_odbc backport is giving me an error (I know that this backport is not "officially supported"

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get

This looks like a fun box... a Voip to Dect gateway, I've mailed them for pricing details.... < <http://www.computex.com.tw/news_archive_detail.asp?index=4053> http://www.computex.com.tw/news_archive_detail.asp?index=4053> Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen E: <mailto:info@betel.nl> info@betel.nl The Netherlands W:

I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX Enterprise R9.0. As EuroISDN it works fine. However, I need to move to QSIG because of a firmware upgrade on the Alcatel PBX which doesn't support EuroISDN (please don't ask why). Besides, I've read somewhere that 2 B Channel Transfers "should" work with * 1.4, the latest 1.4 libpri and QSIG. So this

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and