similar to: SIP Extensions and loss of Internet connection

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, all! I'm being based on this document [1] to send and to receive calls using ekiga.net. But I'm seeing, in an Asterisk console, several messages of this type: [Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout: -- Registration for 'danib2 at ekiga.net' timed out, trying again (Attempt #4775) -- Got SIP

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend. I'm getting: [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --------------------------------------------------------------------------- 400/400

Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and

Hi all! Is it possible to use a usb printer from a DomU? I don''t get to see any device using both lscpi and lsusb. I''m using Xen 3.0.2-2. Thanks in advance. Regards, Daniel -- Daniel Bareiro - System Administrator Fingerprint: BFB3 08D6 B4D1 31B2 72B9 29CE 6696 BF1B 14E6 1D37 Powered by Debian GNU/Linux Etch - Linux user #188.598

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Days ago I bought a OpenVox A400P card with a port FXS and another FXO that I am testing with my Asterisk installation in my house. I'm using Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on Debian GNU/Linux Lenny. I was reading "The future of telephony" and this [1] document looking for information about

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 compiled by myself with the source code of the official site of the project. I would like to update to one more newer version. I suppose that the recommendable thing is to maintain me in branch 1.4, reason why in this case it would be 1.4.30 that I suppose that

Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.fuller at

Folks, I cannot seem to be able to place a call from a dialup connection (this is the first time I try to do this)

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems,

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

Say, I just picked this up on my messages! There are a whole host of these requests! Anyone know whow there people are? Is there a way to report them? Any suggestions as to how to block them? [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '"912" <sip:1 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010]

Hello, I am using Asterisk 1.4.24.1 version in production. OS is Centos 5.3 64 bit & RAM is 8 GB. I am facing crash in asterisk approx each 12 hour. When it crashes I see below linesin asterisk logs. [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( I debugged asterisk source code in details & I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I had the opportunity to test a Sangoma A200 card and I have some doubts that I would like to consult: I tried to detect the card and I had no success using the wctdm module with DAHDI. I guess this is because electronics is different because the TDM400P and OpenVox A400P cards have separate modules for each channel, while the Sangoma A200

Hello I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora 5 for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card, so no need for zaptel and libpri), but I'm stuck: The GrandStream BudgeTone phone fails registering with Asterisk :-/ Following the "Asterisk - The Future of Telephony.pdf", here's what I did: 1. Installed Fedora 5,

Hello everyone and welcome to my first post to the list! After studying for a couple of weeks, I finally built * for the first time last night, and of course had the same SIP-behind-NAT woes that plague all of us who use NATted connections. It was therefore with no small joy that I read the fix for that that Walter Snel proposed (q.v.:

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

When run asterisk ?vvvgc IT show me this error Asterisk Ready. *CLI> Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'phone@192.168.0.6' timed out, trying again Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from '<sip:phone@192.168.0.6>' failed for '192.168.0.6' Apr 11 08:59:27 NOTICE[81926]: