similar to: Echo and static on PRI with errors.

Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P

I am having a problem with and Asterisk installation where two ports connected to a TDM400 card will have a very loud noise when you try to dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank 8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18. The problem always happens with two ports (34 and 35) which are connected to two GSM gateways. They will work fine for a week

Hi, I have problem using Asterisk.I have isdn-pri and openvox d110p card in my computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all pins to the isdn done by telco workers). I got green led on isdn which is sign that isdn is working and that is connected to openvox, right ? I compiled newest versions of libpri zaptel and asterisk and had no problems during that. When I started

Hi all, So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my

Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a "straight" cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same.

Hey Everyone, I would like to start testing/playing with PRI channels but I don't have access to a PRI line. Is it possible to do the equivilent of a crossover between two PRI Cards (say Digium's TE120P)? What I was thinking is that I could set one asterisk box up with a PRI card set as the TE and provide clocking and another box exactly the same but with the card setup as NT. I think

Hello Team I have installed the new DL580 and used the new TE420B to add capacity on our ivr. Before I put new E1?s I decided to first move the old e1 from the old system to this new one but it has errors which not only affect the audio quality, but also cause the asterisk to refuse any call after sometime even though the channels seems up and active {seems d-channel fails}.. When processing

The Asterisk Development Team has released Asterisk versions 1.2.33, 1.4.25.1, 1.6.0.10, and 1.6.1.1. The released versions are available at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes a REGAUTH loop related to security issue AST-2009-001. Asterisk release 1.2.33 also addresses a small compile time error in chan_spy. For more information about the security issue,

The Asterisk Development Team has released Asterisk versions 1.2.33, 1.4.25.1, 1.6.0.10, and 1.6.1.1. The released versions are available at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes a REGAUTH loop related to security issue AST-2009-001. Asterisk release 1.2.33 also addresses a small compile time error in chan_spy. For more information about the security issue,

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command

I've * in a small office with 10 internal sip extensions on aastra's. Outgoing is pstn over dahdi, voip over teliax and iax to another office. This morning no calls could be made: iax to branch offfices, voip iax over teliax, pstn, or even internal extensions. The aastra's showed "Not in Service". A "core restart now" got everything working again. Before I

Hi, I've noticed that my MeetMe install seems to think chan_dahdi is missing: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) However, it definitely is since I have 3 PRIs functioning normally :) Is there anything I should check before I restart asterisk this evening to see if that fixes it? Thanks. -- James ** Please CC me on

Hi, I have a PRI problem where it appears that my system is not responding to SETUP messages on a channel. It seems to be retransmitting a significant number of RELEASE messages to clear a call that is most likely to be long gone. This causes a huge issue because I get a bunch of hangup cause 102s (timeout). I'm using a TE410P (1st Gen) as my PRI card. Has anyone seen this at all? Thanks.

Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear the problem seemed to be to restart Asterisk. It appears to start after the following message

For our internal fax machines, I'm checking if the faxes are going to branch offices. If they are, I want to capture and email them to the branches. I've set up extension 8447 to test this. A fax machines is connected via an SPA 2102 on 173. Any calls from 173 are sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than "NOT_INUSE". I have two extensions: 6666 and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use 6666 to call 6668 and in the dialplan have a noop to see what

Hi All, I ' m using Cisco 7970 IP Phone and Asterisk 1.6.0.10-FONCORE-r40 (Tirxbox). My problem is that I upgrade my phone to SIP image but now this phone is not registering. The error likes this : SIP/2.0 403 Forbidden (Bad auth) The phone and Trixbox are in the same network. There arenot any NAT rules. Can you help me please?

Hi all, I've just built a new installation of CentOS release 5.3 (Final) and have installed both <http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz>Asterisk 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing complex - Pentium Dual core 2ghz - 1gb ram - 70gb

Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue

Hi, I'm seeing a very strange error when dealing with Diversions. If a call setup to a number comes to an Asterisk server, that server sends a request to a third proxy, that proxy sends the call back with a Diversion flag, Asterisk complains about the host not existing (and the host is the number). Here's the output from the Asterisk CLI with SIP debugging enabled: <--- SIP read from