similar to: fix isn't

Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capability; 3921,3922d3919 < p->capability = user->capability;

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

hi ppl, i get this error in my asterisk CLI: Aug 31 02:45:31 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register Aug 31 10:22:24 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register I guess it is related to my problem i have with one of my voip providers: i'm using asterisk 1.2

Hi, I have an * server on a routable (public) IP address and a sip client behind NAT using a Grandstream phone. He is connected through a bi-directional satellite so he has a bit of latency involved. Usually I can dial this extension and them to me. But I keep getting a registration failed message. I have other sip clients not on a satellite and they don?t get these time outs. So I assumed it

We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA

Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and

Hi, I am using SIP extensions connected to the PSTN with the CAPI Channel driver. All works fine except that one of the sip phones keeps ringing when the caller hangs up before extension is answered. The phones are grandstream 100, though we get the same behaviour using other phones (X-lite, Kphone). It behaves the same regardless of whether the incoming call is from a SIP extension or an

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: > > - with a SIP phone configured as 192.168.1.190, and with its SIP > > server being 192.168.1.190 > > That doesn't look right. Do you have another "SIP

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

I have 2 Grandstream HT-286 devices and an Asterisk server. The * Server is not using NAT and has port 5060 opened up. One HT-286 is using traditional NAT and the other HT-286 is behind a residential DSL router/firewall. I have the HT-286 setup as the "DMZ Host" in the router/firewall so that all incoming connections are forwarded to the HT-286. HT-286-1 ====== NAT FW ====== * Server

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hello, long time lurker, first time writer.... We have the following set up ITSP | | Internet | | Cisco 2600 | | Switch----Asterisk Server running 1.0.9(has public ip) | | Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat) | | Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9) The grandstream registers with the public asterisk server fine. I even

Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to

Hello All, I have gone thru all the resources I could find on google on asterisk + iconnect and managed to get outgoing calls working. However, I cannot get incoming calls to work at all. With the sip debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would

Hey there i'm trying to get an Asterisk 11.11 with encryption working with my Grandstream phones. But i stuck. To avoid NAT problems i'm using IPv6 Just with TCP/TLS it's working fine. Only the SRTP funktion is not working. Asterisk tells me WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fa10800f5a0 (len 681) to [2a02:1205::...]:37635 returned -2: Success and also SSL

Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both