similar to: SIP peer registration interval - SOLUTION

On Thu, 17 Feb 2005 15:04:50 +0100 Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I'm registered with sipgate, a German SIP provider. >Configs works fine so far. Trouble is, after a while, it >seems, my registration is dropped by sipgate. How do I >tell * the interval for * registering with a provider? I >suppose, the re-registration

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Which phones do you have? We are using Cisco 7940G phones and I have been able to do this by modifying the dialplan.xml for the phone to rewrite numbers as they are dialed to include the "9" in front of whatever is dialed from the phone. Now you can use the received calls menus without having to edit the numbers before hand. Calvin On Jan 29, 2005, at 12:13 PM, Stefan Gofferje

Hi all I use sipgate and FWD but seem not to get it going. I do not have NAT on the asterisk box (static ip). The asterisk box has 2 network interfaces. One internal and one external. Now when I make an call to a FWD or SipGate number all I get is -- Executing NoOp("SIP/113-6d2e", "") in new stack -- Executing Goto("SIP/113-6d2e",

Hi, Using a couple of sip phones and using asterisk to connect them to a single sipgate.de account. if I call a mobile I have no problem makeing conversions. If the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden

Hi everybody, I have strange problem: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222

Hey, we're experiencing a weird problem with Asterisk 1.8.13.1 (1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via a PBX (sipgate.de) work perfectly fine, almost 100% of the time. However, calls that are routed to sipgate.de, which then routes the call back to our Asterisk instance are "silent" most of the time. What I mean with that is that even though RTP traffic

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for 'whatever@provider.tld' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK

Hi list! I?m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can?t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742

Hi Everyone, I'm using Asterisk, actually A@H 1.1 with all Grandstream 102 phones. NAT is not an issue as all including the server have public IP's The problem is that the phones keep losing registration with the server. I have not timed this exactly to see if they drop off with exactly the same frequency. The SIP TRUNK connection to my provider SIPGATE does not lose registration, and

HI i am trying to register mgcp gateways(Polycom 501, 601) to asterisk as a call agent, mgcp gateways are not registering to the call agent. Please help me on this if any one knows how to congigure the mgcp.conf on asterisk as well as an MGs. The following are the details of mgcp.conf on asterisk. mgcp.conf [general] port = 2427 bindaddr = 0.0.0.0 [0004f205c258] //MG MAC Address host =

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 On 06/14/2015 01:48 PM, Luca Bertoncello wrote: > Guenther Boelter <gboelter at gmail.com> schrieb: > >> Don't use Port 5061, your SIP-port should be always even like >> 5060, 5062, 5064 or 5066. > > Could you please explain why? I see in /etc/services, that 5060 is > the port for SIP and 5061 for SIP-TLS, but

Hi, I have researched more into the problem of my Asterisk set-up not answering calls. The following error was shown on the CLI, can anyone explain what the problem causing Asterisk to not answer the SIP calls be? Information: I have an Asterisk box on a home LAN, behind a D-Link router/firewall connected to a cable modem. The 82.x.x.x is the IP for my cable modem. 192.168.0.101 is my

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station