search for: asterisk3

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2012 Sep 17
1
iax2 trunks between asterisk servers
...t displayed if the called party is on a different server, it works if the called party is on the same server. On each server sip clients show calleeid on calls but if the caller is between servers calleeid doesn't work. Callerid is working fine though. A call from a SIP client on asterisk2 to asterisk3. All phones are snom 760s. Any ideas or suggestions appreciated. iax.conf (asterisk2 10.6.1) [general] bandwidth=high allow=all shrinkcallerid=no [asterisk3] type=friend username=asterisk2 secret=secret host=10.101.0.3 context=incoming sendani=yes trunk=yes iax.conf (asterisk3 11.0.0-beta1...
2006 Jun 08
0
Problems with IAX
Hi, Here's my setup: (PSTN)--[ASTERISK1]--(IAX)--[ASTERISK2]--(IAX)--[ASTERISK3] I don't run asterisk 1, but I do run asterisk 2 and asterisk 3. I have a DID via PSTN on asterisk 1 that is directed at asterisk 2 via IAX. On asterisk 2 I want to direct that DID at asterisk 3. I have done so, I have the IAX stuff setup between asterisk 2 and asterisk 3, but when...
2007 May 30
2
(no subject)
Need some help with IAX trunking. I've got six systems: AsteriskM (main) ___________________|____________________ | | | | | Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5 AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk boxes are using ztdummy for timing, they are all using IAX trunking. My calls come in over Sip or Zap to asteriskm and are routed to one of the asteriskN servers based on load. The routing is done by a sm...
2006 Jun 16
5
asterisk load balance
...ingle server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--------------------------+ DB and file server +---asterisk2-----------------------+ +---asterisk3-----------------------+ Does anyone has load balancing experience implemented in asterisk that can share? Does my design work? Does any conflict will happen in my design? Any comment? Thanks!
2015 Jan 26
2
asterisk 11.14 - voicemail incorrect duration
Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] --
2020 Apr 30
2
SIP TLS not working, Asterisk 16.9.0
Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem
2015 Sep 16
4
Realtime Voicemail MWI
Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without
2010 Nov 05
1
Asterisk 1.8 Installation Problem
Hi, We want to upgrade both our servers to asterisk 1.8, the one from Romania and the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on a test machine running CentOS 5.5 with the kernel: Linux asterisk3 2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686 i686 i386 GNU/Linux . I`ve tried many things from the forums and mailing lists but none seemed to help me. Our problem is that when we want to compile asterisk 1.8 we get this error: /packages/asterisk-1.8.0/addons/chan_ooh323.c:3888...
2006 Jun 20
8
fail to make call
Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register infomration. UA1 can make call to UA2,UA3 and UA4. UA2 can make call to UA1...
2005 Jan 13
2
Looking for a wireless phone... wifi ortraditional wireless ?
...49 PM > >> Subject: Re: [Asterisk-Users] Looking for a wireless phone... > >> wifi or traditional wireless ? > >> > >> > >> > >> An unflattering zyxel review: > >> > >> http://slacker.com/~nugget/asterisk3.php > >> > >> I can't help but think my questions are out of place on this > >> list... I'm > >> asking questions about SIP phones and everyone else is talking > >> about > >> asterisk. Sorry. &gt...
2015 Jan 26
0
asterisk 11.14 - voicemail incorrect duration
...nique On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: > So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only > count 2. What can be the reason? It is not silence. Are you sure? The value for silencethreshold (140) is unusually large. -- Stefan Tichy ( asterisk3 at pi4tel dot de )
2015 Jan 27
1
asterisk 11.14 - voicemail incorrect duration
Hi Stefan, Stefan Tichy <asterisk3 at pi4tel.de> schrieb am Mon, 26. Jan 23:56: > Hi Dominique > > On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: > > > So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only > > count 2. What can be the reason? It is not silence. > &gt...
2020 May 01
0
SIP TLS not working, Asterisk 16.9.0
...1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the system default settings at the bottom of "/etc/ssl/openssl.cnf", restart asterisk and try again. Keep in mind that this will affect the whole server. -- Stefan Tichy ( asterisk3 at pi4tel dot de )
2023 May 24
0
Problems Solved, two left
...more complicated. You might have to change the phone configuration. > [yealink] > type = aor > contact = sip:Steve at 192.168.1.185 There should be no "contact" parameter for a phone. The phone sends the required information with the register request. -- Stefan Tichy ( asterisk3 at pi4tel dot de )
2004 Jul 21
0
extensions.conf variable declaration
Hi, I'm setting up multiple asterisk servers and trying to do the classic DIAL(IAX2/asterisk1/${EXTEN}&IAX2/asterisk2/${EXTEN}&IAX2/asterisk3/${EXTEN},15) After googling a bit, I fell on a discussion about putting this in a variable so that adding additionnal servers would be easy. I can't seem to find the link anymore, but it went something like this: extensions.conf: [global] SERVERS = IAX2/asterisk1&IAX2/asterisk2 [defau...
2007 May 16
0
Passing dialstatus back through an IAX chain ..
I feel I'm doing something obviously wrong here and will kick myself when I see the answer!!! The scenario: SIP phone -> Asterisk1 -> IAX -> Asterisk2 -> IAX -> Asterisk3 -> PSTN So I place a call from the SIP phone. A1 picks it up and forwards it to A2 which forwards it to A3. A3 sends the call to the PSTN. I control A1 and A2, but not A3. When a call fails (for either unavalable or busy), A2 sees the failure code back from A3. A2 doesn't do anything wi...
2014 Jan 15
1
How to tell Asterisk to to send Ringing signals as into RTP
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
2014 Nov 09
0
taskprocessor fails to allocate memory
...istener for taskprocessor 38660bf7-eec2-4ce6-a9d7-63c8178a0556 [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory After 18 instances of Asterisk using parameter -C /etc/asterisk1/asterisk.conf -C /etc/asterisk2/asterisk.conf -C /etc/asterisk3/asterisk.conf etc. The machine has 180 GB of RAM and 16 cores. ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 1048576 max locke...
2014 Apr 30
2
Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip reload" or "iax2 reload" does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working
2023 May 23
3
Problems Solved, two left
And I think they're both small. Solved: tcpdump showed no packets coming in, so I went to my DID provider's Website to discover to my intense embarrassment that the DID number had been set up forwarded to their voicemail. I got egg on my face for this one. I changed that setting to SIP/IAX and packets now arrive and go where they should. Two problems remain. 1. Still can't