similar to: Client Asterisks can't connect when main Asterisk reboot

Displaying 20 results from an estimated 8000 matches similar to: "Client Asterisks can't connect when main Asterisk reboot"

2003 Oct 14
3
*/SER/FW
Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan
2003 Aug 17
2
Recomendations for an ISDN-PBX to use with asterisk
Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus
2012 Aug 20
1
Asterisk as TLS server as well as TLS client
Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer
2006 Nov 07
3
connect Sipura with Asterisk - both behind NAT
Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph
2008 Dec 20
2
how to get /var/run/asteris/asterisk.ctl
Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.
2009 Jun 18
2
snom mass deploy help
Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:
2006 Oct 26
10
ECHO Cancellation in SIP Calls
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the
2020 Apr 21
3
Dialplan - using multiple AND or OR in set is it possible ?
Hello, we want to use something like same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) Problem is that result gives C=1) & Set(D=2) & ... Is there a possibility to use multiple AND or OR in such a way ? -- Daniel
2004 Jan 09
1
Asteriks as SIP<>H323 Proxy?
Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,
2004 Apr 09
5
vm e-mail notification stopped
After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm
2017 Apr 19
2
IAX2 getting stuck
On 4/19/17 4:09 PM, Antony Stone wrote: > On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: > >> I have a server that had been operating for a few years now with >> IAX2 trunks to several other servers. Since yesterday all IAX2 trunks >> now say UNREACHABLE. > ...snip... > >> So far the only thing different is that the receive queue for port
2017 Apr 19
2
IAX2 getting stuck
On 4/19/17 4:23 PM, Antony Stone wrote: > On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote: > >> On 4/19/17 4:09 PM, Antony Stone wrote: >>> On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: >>>> I have a server that had been operating for a few years now with >>>> >>>> IAX2 trunks to several other servers.
2017 Apr 20
2
IAX2 getting stuck
If SIP goes to the same provider then yes. Still I would check a packet capture for better understanding. BTW, did you try iax debug? ??, 20 ???. 2017 ?. ? 19:46, Carlos Chavez <cursor at telecomab.mx>: > On 4/20/17 12:45 AM, Kseniya Blashchuk wrote: > > Can it happen that the routes lead the traffic through another interface? > Did you try a packet capture with tcpdump? Do the
2017 Apr 20
2
IAX2 getting stuck
Can it happen that the routes lead the traffic through another interface? Did you try a packet capture with tcpdump? Do the packets really leave the usb adapter? Can asymmetric routing be in effect? Maybe there were some static routes that disappeared when the adapter was unplugged... On Thu, Apr 20, 2017, 12:41 AM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On
2004 Aug 11
7
H323 call dropped when answered
Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.
2015 Feb 17
2
Res_fax - FAXOPT(faxdetect)
Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number.
2015 Feb 18
2
Res_fax - FAXOPT(faxdetect)
I solved the issue by not answering the call as I assume others have done. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le
2006 Jan 24
1
need help asterisk and AS5300
hi All Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ? i need informations sample config for that, or can show how to route docs . thanks Dirgan --------------------------------- Meet your soulmate! Yahoo! Asia presents Meetic - where millions of singles gather -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Jan 25
1
dialplan and "*"
Hi, I'm analyzing freepbx extensions. When creating ivr with freepbx, it writes like this: exten => 1111,1,Answer exten => 1111,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID) exten => 1111,n(USERCID),Macro(user-callerid,) exten => 1111,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME}) exten =>
2007 Aug 17
1
gsm errors
Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions ram -------------- next part -------------- An HTML attachment was scrubbed... URL: