similar to: can't get pap2 to register from outside the LAN.

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes.

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

Hi, I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001

Been having a problem with my two Sipura 2000's dropping calls from the SPA-2000 side. Seems the calls are dropped right before the "Next Registration" time. Calls drop about ever 60 minutes or so. I have dialed from one port to the other and let it sit. After about 60 minutes or so the calls get dropped. System details are below Asterisk ver. CVS-HEAD-11/27/04-23:42:45 RHEL 3

Maybe I just woke up too early today. I have SJPhone and X-Lite working perfectly but I cannot for the life of me get sipXphone working properly with Asterisk. Its probably something stupid on my part, but does anyone have a quick setup sheet for it? -Kerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit : > Hi, If you're not using RTP encryption did you uncheck the option in your RTP TAB from identity ? > > This is the log. ex dialling 0 from snom phone > > > <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 > <http://123.231.72.210:33878> ---> > INVITE sip:0 at 54.206.59.252

On 23/03/2020 18:51, Joshua C. Colp wrote: > On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com > <mailto:john at calva.com>> wrote: > > > > Why is asterisk giving an error 500? I can find no reason, there > is nothing in any log. > > > The sequence number is from the past. The first SUBSCRIBE is sequence > number 22 (check the

I recently got a SNOM 360 and have been trying to get the extension lights to work. I can see the subscriptions with sip show subscriptions but I don't see any notifies when a call is made. I must be missing something because I've tried looking to see if anyone else has had this problem but the only solutions I've seen have been to put hints in and I have those. Any suggestions?

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

So I've got a bit further with my  project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

Hey, I'w got a problem (bug maybe?). I have recently got my Asterisk to work perfect and I'm not trying to setup some dial routes and get the system working as I wan't it to. Yesterday I was installing Festival and also did a "aptitude upgrade" on my Debian Unstable installation. After that the problem started. After some serious testing yesterday night and today I have