[asterisk-users] Rejected calls to Sylantro server
I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register against a Sylantro server in front of a Metaswitch. I'm able to register and receive inbound calls but outbound calls are rejected by the far end. The username and password have been checked repeatedly. Putting the same authentication and server IP into a softphone or polycom phone work fine for inbound and outbound calls. Has anyone made this work in the past? This is the rejection sent by the switch at the other end: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.1.4.110:5060;received=67.55.x.x;branch=z9hG4bK71d476d2;rport=5060 From: 515XXX sip:[EMAIL PROTECTED];tag=as6a5cd6c8 To: sip:[EMAIL PROTECTED];tag=aprqngfrt-n2bk9j1c6 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
On Mon, 24 Apr 2006, Rich Adamson wrote: Alexander Lopez wrote: I was once told by a lineman that the cables they use didn't have that many twists in them because it wasn't needed, and that the extra twists would effectively use more cable and thus cost and weigh more than triple what they do now. Good thing he doesn't work for a cable manufacturer as that's a total crock of crap that even an inexperienced person should be able to detect. (You can't twist two wires to make them weight three times as much, or cost three times as much.) A foot of cat5 has more than 12 on each of the individual wires inside. Not much but there is some difference. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tsu-600
On Sun, 26 Mar 2006, mike webb wrote: i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ?? The Adtran TSU-600 can be made to work like a normal channel bank. I'm not using one with Asterisk, but the Adtran docs are very well written. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk featdmf signalling.
On Mon, 9 Jan 2006, Michael Baird wrote: I've recently started PIC'ing some calls into a asterisk box across a feature group D trunk from Verizon. Everything seems to work ok, except for some reason Asterisk doesn't grab the full caller ID from Verizon. I can see that they do send it, but Asterisk drops the first 2 numbers. Looking at the debug log I see. I see that Verizon is sending the digits and the asterisk debug seems to understand it fine. But within Asterisk it appears to truncate the first two digits of the callerid. The called-from appears in both the .csv log and my mysql cdr log as 72652437 and it sends this number out as the callerid as well. I'm at a loss as to why it misses the first 2 digits, any suggestions would be appreciated. There are two kinds of FGD protocols on Asterisk, I got stuck with this when it only supported the one that my IXC didn't support. Read through chan_zap to see to see the format of each one. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk featdmf signalling.
On Mon, 9 Jan 2006, Michael Baird wrote: On Mon, 2006-01-09 at 14:20 -0600, Dave Weis wrote: On Mon, 9 Jan 2006, Michael Baird wrote: I've recently started PIC'ing some calls into a asterisk box across a feature group D trunk from Verizon. Everything seems to work ok, except for some reason Asterisk doesn't grab the full caller ID from Verizon. I can see that they do send it, but Asterisk drops the first 2 numbers. Looking at the debug log I see. I see that Verizon is sending the digits and the asterisk debug seems to understand it fine. But within Asterisk it appears to truncate the first two digits of the callerid. The called-from appears in both the .csv log and my mysql cdr log as 72652437 and it sends this number out as the callerid as well. I'm at a loss as to why it misses the first 2 digits, any suggestions would be appreciated. There are two kinds of FGD protocols on Asterisk, I got stuck with this when it only supported the one that my IXC didn't support. Read through chan_zap to see to see the format of each one. Thanks for the reply, I've tried both types, the other type SF_Featdmf doesn't work at all. So I'm pretty sure I am using the correct signalling specification. I came to that conclusion using log statements, if it's not making it to the dialplan I would investigate in the channel driver either by reading it or putting in log statements in strategic places. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Lucent TNT w/11.0.2
On Sat, 5 Nov 2005, Shane DeRidder wrote: I've been scouring the mailing list archives for an answer to this, and cannot find one. I'm hoping someone else out there has run into this. Communication between the TNT and Asterisk seems to be operating properly, but I'm unable to accept or originate calls. When I attempt to dial out, I see the following in the TNT's syslog: 10.0.0.10 = TNT 10.0.0.103 = Asterisk new MEDIA-GATEWAY set name = voip set active = yes set protocol-type = sip set mg-sig-address type = specific set mg-sig-address ip-address = 10.0.0.10 set mg-rtp-address ip-address = 10.0.0.10 I have these set to what would be 10.0.0.103 on my TNT and it's working. dave set transport-options type = udp set transport-options heartbeat = yes set voip-options codec-options g711-ulaw dtmf-tone-passing = rtp set voip-options codec-options g711-ulaw silence-det-cng = yes set sip-options primary-proxy ip-address = 10.0.0.103 set sip-options primary-proxy transport-options heartbeat = yes set sip-options registration-proxy ip-address = 10.0.0.103 set sip-options unknown-ani = 00 set sip-options unknown-name = Unknown set sip-options blocked-ani = 00 set sip-options blocked-name = Blocked write -f My 12 T1/PRI are configured exactly alike: new T1 set name = PRI-0 set physical-address shelf = shelf-1 set physical-address slot = slot-1 set physical-address item-number = 1 set line-interface enabled = yes set line-interface frame-type = esf set line-interface encoding = b8zs set line-interface signaling-mode = isdn set line-interface default-call-type = dnis-or-voip set line-interface switch-type = nat-isdn-2-pri set line-interface front-end-type = csu set line-interface channel-config 24 channel-usage = d-channel set line-interface collect-incoming-digits = yes set line-interface voip-gain-control output-pad = 9db-loss set line-interface media-gateway = voip set line-interface egress-ani-dnis-format = dnis write -f Asterisk sip.conf: [maxtnt] type=friend host=10.0.0.10 dtmfmode=inband callerid=MaxTNT maxtnt context=toll-access qualify=yes reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw [xxx] type=friend host=dynamic nat=yes callerid=Name xxx context=toll-access dtmfmode=info call-limit=1 [EMAIL PROTECTED] disallow=all allow=g729 allow=ulaw Asterisk extensions.conf: [toll-trunks] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _1NXXNXX,2,Hangup [local-trunks] exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _NXX,2,Hangup [local-access] include = extensions include = local-trunks [toll-access] include = local-access include = toll-trunks I apologize if this is considered off-topic. My thoughts are that I have a problem with the configuration of my TNT and not Asterisk itself. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP service from Free Téléc om
On Sat, 17 Sep 2005, Stewart Nelson wrote: I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free Télécom comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO interface. However, to improve quality, reliability, control, etc., I'd like to have Asterisk directly access the underlying MGCP service. Somewhat off topic, who makes the hardware that Free is using? Is it available/being used in the US? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Adtran TA 616
On Thu, 8 Sep 2005, Nick Colton wrote: Has anybody had any luck getting an Adtran Total Access 616 working via the Ethernet port/MGCP to an * box? The voice lines don't seem to be coming up and I wasn't sure if I had something missing. I had tried to get it working a few months ago but didn't get anywhere either. It wasn't clear if the mgcp stack could work over either interface or only over the wan side, so I set it up back to back with a cisco running to my asterisk server, but never got anything going. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TNT and SIP problem
On Mon, 25 Jul 2005, Eric Wieling aka ManxPower wrote: Leon Sun wrote: Try to use like following [tnt] type=friend context=fromtotnt dtmfmode=rfc2833 host=XXX.xxx.xxx.xxx I am using this way. You do realize that host= only applies for calls from Asterisk to the TNT, right? You need permit/deny to match for inbound connections. Your Asterisk server is open to anyone that claims to be the user tnt. Thanks for the pointer. There is no sample in the sip.conf file that mentions permit and deny. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TNT and SIP problem
On Tue, 26 Jul 2005, Kevin P. Fleming wrote: Eric Wieling aka ManxPower wrote: Thank you for the correction. Is this new to CVS-HEAD, or does it apply to 1.0.x as well? 1.0.x sip.conf.sample only lists the host option for a peer, not for a user. It only applies to 'peer' entries, but the important point is that 'peer' entries _are_ allowed to place calls through Asterisk, not just receive them. That applies to both 1.0.x and CVS HEAD. In CVS HEAD _every_ config option allowed on a 'user' entry is also supported on a 'peer', although that's not the case in 1.0.x (and won't be changed there, for obvious reasons). Realistically, this means that 'type=user' and 'type=friend' are of little value any more in CVS HEAD, except for some unusual cases. I did have it set up as friend, but I see now why that didn't work. Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TNT and SIP problem
On Mon, 25 Jul 2005, Leon Sun wrote: Did you enable media profile and put Asterisk IP into proxy field? Use lines to check you TNT Read media default List sip-options Yes, my asterisk server is in primary-proxy and registration-proxy. There is a trusted-proxy, should that be set? The TNT is still logging 407 errors, but I have it dumping into a context with a _. extension. dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Sunday, July 24, 2005 12:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TNT and SIP problem I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid=MaxTNT maxtnt context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing [EMAIL PROTECTED]: Calling = 201,Called = 2700674, Q850 Cause = 21,Sip Response = 407 (Proxy Authentication Required),Progress Cause = NONE Jul 24 14:55:12 tnt1 1/2: [1/2/11/0] STOP: ''; cause 821.; progress 1407.; host 0.0.0.0 [MBID 11; 201-2700674] Jul 24 14:55:12 tnt1 1/1: [1/1/3/1] Far End Hung Up, External cause code 021 I just have a T1 port from the asterisk machine cabled to the TNT with a T1 crossover trying to send calls out of the asterisk machine via T1 and back in via SIP until the PRI's are turned up. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TNT and SIP problem
On Mon, 25 Jul 2005, Leon Sun wrote: Try to use like following [tnt] type=friend context=fromtotnt dtmfmode=rfc2833 host=XXX.xxx.xxx.xxx [maxtnt] context=fromtnt type=peer host=x.x.x.x dtmfmode=rfc2833 callerid=MaxTNT maxtnt qualify=yes reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw insecure=invite ; also tried insecure=very I'll try setting up peer separately tomorrow. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Monday, July 25, 2005 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TNT and SIP problem On Mon, 25 Jul 2005, Leon Sun wrote: Did you enable media profile and put Asterisk IP into proxy field? Use lines to check you TNT Read media default List sip-options Yes, my asterisk server is in primary-proxy and registration-proxy. There is a trusted-proxy, should that be set? The TNT is still logging 407 errors, but I have it dumping into a context with a _. extension. dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Sunday, July 24, 2005 12:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TNT and SIP problem I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid=MaxTNT maxtnt context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing [EMAIL PROTECTED]: Calling = 201,Called = 2700674, Q850 Cause = 21,Sip Response = 407 (Proxy Authentication Required),Progress Cause = NONE Jul 24 14:55:12 tnt1 1/2: [1/2/11/0] STOP: ''; cause 821.; progress 1407.; host 0.0.0.0 [MBID 11; 201-2700674] Jul 24 14:55:12 tnt1 1/1: [1/1/3/1] Far End Hung Up, External cause code 021 I just have a T1 port from the asterisk machine cabled to the TNT with a T1 crossover trying to send calls out of the asterisk machine via T1 and back in via SIP until the PRI's are turned up. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TNT and SIP problem
I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid=MaxTNT maxtnt context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing [EMAIL PROTECTED]: Calling = 201,Called = 2700674, Q850 Cause = 21,Sip Response = 407 (Proxy Authentication Required),Progress Cause = NONE Jul 24 14:55:12 tnt1 1/2: [1/2/11/0] STOP: ''; cause 821.; progress 1407.; host 0.0.0.0 [MBID 11; 201-2700674] Jul 24 14:55:12 tnt1 1/1: [1/1/3/1] Far End Hung Up, External cause code 021 I just have a T1 port from the asterisk machine cabled to the TNT with a T1 crossover trying to send calls out of the asterisk machine via T1 and back in via SIP until the PRI's are turned up. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pass-through
On Wed, 1 Jun 2005, Dustin Wildes wrote: Adam Vocks wrote: In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didn?t know if this could be done with two T1 cards and asterisk? I've done the exact same thing. We had a 23-channel PRI that a client was using for voice, but had a small IVR for their banking application that had direct analog lines pointed to it. I ordered an Adtran Total Access 750 and an additional T1 (T100P) card. The TA750 had 24 analog lines, with one T1 interface. The asterisk server had 2 T100Ps one card was for the PRI, the second was a cross-over to the Adtran 750. Works great, don't see why it wouldn't work for you in the same method you are talking about for a modem pool. There is a better device if you have a PRI, an Adtran Atlas 550 is basically a full phone switch. You can put an entire dialplan on it to do router based on DID/DNIS. It will also do channelized T1 to PRI conversion each way. Very slick boxes, I'm about to set one up for another asterisk user to split 1 PRI to 12 pots lines for an older switch, 1 PRI for Asterisk, 1 channelized T1 for a modem bank, and some FXO ports for an older Brooktrout card. If anyone wants more info on them let me know. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Router Terrible Voice Quality
On Sun, 17 Apr 2005, Greg Boehnlein wrote: On Thu, 14 Apr 2005, Rod Bacon wrote: I have been frustrated by a variety of zyxel issues/products and have found the best solution for all of them lies in a cylindrical receptacle that sits beside my desk... I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a built in DSL modem, and a single FXS port. Decent little router, now that the latest firmware is out, but tcp and udp timeouts through NAT seem to be set a little low, so I lose SSH sessions. I bought a dozen and have had bad luck with them. I couldn't keep an ssh session for more than 15 seconds. Trying to update firmware turned two of them into paperweights. I couldn't get the FXS to ever register. Other than that, it looked like a good idea. Is the NAT timeout configurable now? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxo problem
I've got a X100P in a compaq proliant 3000. My system stops taking calls and making calls. I had been getting the FXO PCI Master abort before updating, I am now running a cvs head checkout from a week or so ago. Now I still have the problem but get more error messages: Found a Wildcard FXO: Wildcard X101P Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) FXO PCI Master abort wcfxo: Out of space to write register 05 with 02 wcfxo: Out of space to write register 05 with 03 wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a Any solution? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CIC Code
On Mon, 28 Mar 2005, Jason Miller wrote: Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. I got it working briefly. I had to talk with the switch techs at the other end for a couple hours and modify the source code to reformat what was being sent down the line. Their definition of FGD and asterisk's definition were not the same. It was a nortel DMS-100 or 250 set up for CLASS 4. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CIC Code
On Tue, 29 Mar 2005, Jason Miller wrote:
Do you remember what you actually changed to make it work cause that is the
same switch that I am dealing with myself if I am not mistaken.
Approximately line 1673 in chan_zap.c, it looks like this:
if (p-sig == SIG_FEATD) {
l = ast-cid.cid_num;
if (l)
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), T*%s*%s*, l, c + p-stripmsd);
else
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), T**%s*, c + p-stripmsd);
} else
if (p-sig == SIG_FEATDMF) {
l = ast-cid.cid_num;
if (l)
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), M*00%s#*%s#, l, c + p-stripmsd);
else
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), M*02#*%s#, c + p-stripmsd);
} else
The switch tech told me that the 00 indicates operator assistance on their
switch, he thought 01 or the 02 in the next line was more correct. Also,
the * and # aren't called * and #, I found out that the switch tech laughs
at you. :-) They are KP and ST.
One strange thing is that asterisk seemed to be putting pauses in the dial
string. In zaptel.c on line 2502 I inserted a printk to look at the final
dial string. There were pauses inserted in the dial string. It would be
nice if monitor could get audio during the dialing instead of just after
answer.
dave
From: Dave Weis [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tue, 29 Mar 2005 06:23:35 -0600 (CST)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] CIC Code
On Mon, 28 Mar 2005, Jason Miller wrote:
Has anyone ever setup Asterisk to pass Feature Group D access while using a
CIC code for outbound calls? If so can you please email the configuration
you have done? I have tried to get this up and running but with no luck. I
have also contacted support and I cant seem to get this going.
I got it working briefly. I had to talk with the switch techs at the other
end for a couple hours and modify the source code to reformat what was
being sent down the line. Their definition of FGD and asterisk's
definition were not the same. It was a nortel DMS-100 or 250 set up for
CLASS 4.
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--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gradual and silent
encroachments of those in power than by violent
and sudden usurpations.- James Madison
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Re: [Asterisk-Users] GR-303 from Central Office supported?
On Wed, 23 Mar 2005, Kevin P. Fleming wrote: Rich Adamson wrote: I'm a little confused on whether the GR303 support in * will accept calls from a Siemens central office that has GR303. I don't know for sure (sorry for responding anyway), but I believe that Asterisk's GR-303 support is the 'network' end only, so that it can control access concentrators. That would mean that is does not have the ability to act as an access concentrator. From zapata.conf in a recent cvs checkout: ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side I haven't tried it but want to sometime. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Strange text on Asterisk console
On Mon, 28 Feb 2005, Tony Mountifield wrote: In article [EMAIL PROTECTED], Carlos Chavez [EMAIL PROTECTED] wrote: On Mon, 28 Feb 2005 20:58:48 + (UTC), Tony Mountifield wrote I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do asterisk -rv on a normal login, either via the console or an xterm, the text appears correctly. No, when I installed the OS, I selected only English(Great Britain). Only some of the characters appear to be Cyrillic. The point is, only the coloured text is affected: the normal text is OK. Anyone actually seen this behaviour before? Yes, it usually happens when there is colored printing to the console if I'm not on that one. I think it prints correct when I have the console active, though. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran Total Access MGCP Config?
I've never set up an mgcp device before. I have an Adtran IAD with the MGCP firmware on it. I have it configured in mgcp.conf like this: [general] port = 2427 bindaddr = 0.0.0.0 [adtran] host = 192.168.2.2 context = default canreinvite = no line = aaln/1 line = aaln/2 The device is configured like this: MGCP Configuration | Standard MGCP 0.1 / NCS 1.0 MGCP Endpoint Config| MGC Address 192.168.1.253 | Local Address 192.168.2.2 | MGC UDP Port 2727 | Local UDP Port2427 | ADPCM Coding IETF (RTP) | RFC 2833 RTP Payload Type 94 | DSCP Signaling0 | DSCP RTP Traffic 0 | Advanced Config [+] I can ping between the devices fine. Doing an mgcp audit endpoint aaln/[EMAIL PROTECTED] gives retransmitting errors. tcpdump shows traffic over the wire. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI in the US?
On Fri, 4 Feb 2005, Joe Greco wrote: OK, I asked this about a week back and met with no repsonse at all. But perhaps its worth trying again. Does anyone on-list have * running BRI to their local telco? I'm considering this as an alternative to my TDM400p card. No, and I've been looking for this for a while now. I'm seriously considering running them into a Cisco 26XX with a ISDN VIC BRI card and seeing if that works. This one bit of the puzzle is the big thing stopping us from going to a VoIP solution, which I'd really like to do. I found this but haven't tried it yet: http://f64.nu/isp/atlas/ That is proof that with enough Adtran boxes you can do anything. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium T100P T1 Card
On Wed, 5 Jan 2005, Wiley Siler wrote: Your expanation is correct. The AdTran delivers the FXS on the wall and is being converted from digital. I hope you are correct about the swapout and I will chase this up with ISP again. Originally, they told me that changing my service required making changes upstream and reprovisioning. Now I am beginning to wonder. So the equipment chain would look like this... Plug from NIU - Asterisk PRI Card - Out Onboard NIC to phones and data separately Look at the model number on the bottom of the adtran, if it says TDM it's channelized data, if it says ATM it's (surprise) ATM and Asterisk can't deal with it yet. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
On Mon, 25 Oct 2004, Cirelle Enterprises wrote: - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:21 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | | for what it's worth, we were told to use RJ48C (Std Ethernet Cable) | | RJ48C is *not* standard ethernet cable. The twisted pairs are grouped | differently. Ethernet cables work OK for T1s if they are only 2 or 3 | metres long. Long ethernet cables give high error rates when used for T1s. According to the t1 techs I've been dealing with, a standard ethernet cable will work (we have been using a 3 foot segment approx 1m) In fact the only configuration that lights all lights is a standard ether cable. T1 cross-over doesn't create a proper link and lights on the t100p and smart jack indicate a failed condition. The pairs in an ethernet cable are the same pairs as a t1 cable, they are paired 12, 36, 45, 78. T1 uses 12 and 45, ethernet uses 12, 36. If you have a good cable it should work fine. Going too long of a distance from your t1 mounting may required a higher LBO on each end to compensate for the additional copper. Also make sure you are using 24 awg wire, not the 26 or 28 awg in most patch cables. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P problem with LD T1
I've got a dedicated LD T1 terminating to a DMS100 switch. My outgoing
calls aren't working, on the switch side they see two sets of dialing,
with the first three digits repeating. I've used the sample
extensions.conf modified a bit to remove the 9 and the 1, like so:
[trunkld]
exten = _NXXNXX,1,Monitor(wav,/tmp/dial)
exten = _NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _NXXNXX,3,Congestion
I see outbound dialing going out as g1/515xxx, but when they monitor
they see 515(small pause)515xxx. Any ideas what is causing it?
dave
--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gradual and silent
encroachments of those in power than by violent
and sudden usurpations.- James Madison
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Re: [Asterisk-Users] Selling asterisk-based solutions
On Mon, 2 Aug 2004, David Gurr wrote: I'm curious as to folks experiences in selling asterisk-based solutions. In sales-speak, what are the common compelling reasons to buy? Those are good reasons, but one compelling reason is that it's pretty inexpensive to set up a system. We sell Avaya systems, mostly Partner. They are one of the least expensive systems to buy and install, but it still runs at least $400/station with voice mail. The last asterisk install I did had a couple pots lines, a nufone account for outbound LD, a couple grandstream phones, and was about $1k, not including the server. It didn't take long to set up and I can administer it from my office. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk Sunday News: Off track with 1.0, moving forward
On Sun, 13 Jun 2004, Olle E. Johansson wrote: * NFAS and GR-303 support Any info on this yet, that is very exciting! -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: makering.pl
On Tue, 8 Jun 2004, Simon wrote: We had done 755 and the path is correct. still get error's : bad interpreter: No such file or directory That is usually due to DOS linefeeds. Try to open it in vi and the run :set ff=unix :wq and try again -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Goryachev Sent: 08 June 2004 09:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: makering.pl Also check the first line of the file points to your location of perl. Usually /usr/bin/perl or /usr/local/bin/perl PS, chmod 755 makering.pl to make it executable. Regards, Adam On Tue, 2004-06-08 at 18:52, Tony Mountifield wrote: In article [EMAIL PROTECTED], Simon [EMAIL PROTECTED] wrote: Anyone used this ? I am having a bit of trouble got the right perms on makering.pl . Should that file be somewhere in particular ? use the reccommended command sox inputfile -r 8000 -c 1 -t ul - rate | makering.pl ring1.bin but i get bash: makering.pl: command not found ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What a Difference a NIC makes
On Thu, 27 May 2004, Harry Flink wrote: Long story made short I swapped out the SMC NICs with 3com and the world is right again. I have absolutely no errors in the console and every phone and peer stays available. If you are having problems and connot isolate why, try another NIC. I'm having this very same problem with 3COM 3c905-tx/tx-m NIC. I never thought it might somehow relate to Asterisk. My kernel is 2.4.26 with Debian patches (distro is unstable Debian SID) and motherboard has VIA chipset. I've always had great luck with the intel eepro100 cards. It depends a bit on your machine, it might be related to the via chipset. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell server for asterisk question!
On Thu, 13 May 2004, Jeff Roberts wrote: I second that warning to stay away from the perc raid, I have one that continuously deals me fits. I've got a couple dozen of them and never had any problems. They are running everything from redhat 6.2 to fedora to rhel 3. dave Leo Ann Boon wrote: The TE410P works with the 2650, I had 1 in there for months. One other thing, avoid the PERC RAID. The Linux driver in kernel 2.4 series is not very stable. FYI. Bartosz Jozwiak wrote: I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
On Mon, 19 Apr 2004, Jeremy Hall wrote: This may not be the case in all areas, but in my area with Qwest as well, all exchanges have the test at xxx-9996. For example, my number is in the 208 area code, 459 exchange, so the full number would be 208-459-9996. It is not tied to any specific number, so I can use any exchange local to me such as 323-9996. It may or may not work in your area, so try not to do it at 3:00 AM until you have verified the number. I'm also in a Qwest area, but that number doesn't work here. All of the techs that I have asked gave it to me with no problems. They are shy about the automatic ANI number, however... dave -Original Message- From: Ed Rubright [mailto:[EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? The next question for me is: How do I found out my telco milliwatt test number? I'm in Washington State using Qwest. The way I understand this, I'm to dialup the telco milliwatt test number and adjust the rxgain values using ztmonitor tool until the Max Audio Hit is in the middle of the bar graph for a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk @ home ?
On Tue, 30 Mar 2004, Kevin Walsh wrote: I'm curious as to what a Washington State free phone number is? I live in Washington State(Spokane) and we get our PSTN service from Qwest which is certainly not free! The poster was probably referring to IPKall (http://www.ipkall.com/). IPKall will assign you a Washington State (USA) phone number for free and have incoming calls routed to the SIP address of your choice. I have one and am using it to route calls to my office in England. Must be a CLEC trying to build up reciprocal comp minutes ;-) -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk @ home ?
On Tue, 30 Mar 2004, calvis wrote: What are reciprocal comp minutes? Please explain. In some states, the competitive and incumbent phone carriers bill each other for calls that they terminate from the other. If a Qwest customer calls a Dave's Phone Company customer, I will get a small amount of money per minute for completing the call. Some states are bill and keep, meaning no one gets paid. ISP's loved it when they could set up as a CLEC and put their modem banks on their own switch, because they are terminating all of these calls and getting paid per minute from the ILEC to do so. The FCC let the ILEC's not pay reciprocal comp for those calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Tuesday, March 30, 2004 1:12 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk @ home ? Must be a CLEC trying to build up reciprocal comp minutes ;-) -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom NetEngine 6200 Router
On Tue, 9 Mar 2004, Jeremy Mann wrote: I have a 6200-8 router, it's a T1 router with 8POTS ports online and has a multitude of options for a voice gateway, anyone on the list here know which would be most compatible(if not simply work) with asterisk? Can asterisk serve as an MGCP gateway, or is there another linux solution that can interoperate? It should work, but I think that model is atm only so you would need to feed it from an atm switch. Unless you have a lot of them, it would be more cost effective to use a normal channel bank. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax (off-topic)
On Tue, 2 Mar 2004, Nate Carlson wrote: On Tue, 2 Mar 2004, Darren Nickerson wrote: Check out T38Modem at www.openh323.org Is there any way to actually get that working with Asterisk yet? As far as I can tell, neither of the H323 implementations for Asterisk support the T.38 protocol. Last time I tried that with openh323 it didn't work either. It would get a page in or out and then crash and burn. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wanpipe cards?
On Tue, 17 Feb 2004, Jeff Gustafson wrote: Searching the mailing list didn't get me much info on Asterisk and Wanpipe cards. Are they usable with Asterisk? In the docs they talk a lot about T1 connectivity (obviously), but can they also be used on a PRI line (T1 PRI)? I have an extra S514 card that I would like to use with Asterisk if it's possible. I was looking at this yesterday. They seem to have a zapata compatible driver for the 514 cards. Look through their docs for VOIP. I don't think they are compiled/built by default. I have an older card that isn't supported, so I can't help any further. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] extravagant behavior, nat problem ?
On Tue, 17 Feb 2004, Alessio Focardi wrote: Hello Eric_Doiron, Tuesday, February 17, 2004, 1:36:57 PM, you wrote: E Does it work if you don't re-write the port to 5090, but rather leave it at E 5060?.. E What is the 'ser'? Ser is the sip proxy I'm using to authorize users; at least I have isolated the problem: my asterisk is running on port 5090. But since fixup for sip is 5060 the subsequent rtp stream does not come in, only the control protocol is allowed, by my (allow any) rule. Maybe there is a way to tell pix that sip is 5060 AND 5090 . any idea ? Yes, you can do that. Log in and run fixup protocol sip 5060 5090 then save and exit. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones
On Fri, 13 Feb 2004, John Bittner wrote: Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. I had a similar problem with an adtran TA750 with digits not breaking dialtone. It would come and go, usually working fine right after a restart. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * dialing before line is open?
On Fri, 9 Jan 2004, john lawler wrote: I've had a sporadic problem recently with one of my users on our POTS line. About 1/3 of the time he dials a number (usually from a speeddial on his phone, I think), he'll get some phone company message (from the outside) about how the call could not be completed as dialed or something like that. I had a similar problem with a client. I hooked up a butt set to the line and listened and only heard 4-5 digits being sent. The problem ended up being high loop current on the line. It should be around 20-25 milliamps, but this one was 45. High loop current can cause a variety of strange problems. There is more information on it at www.sandman.com if you want to look. The easiest way to fix it is to put a 500 or 1k ohm 1 watt resistor in series with both tip and ring. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 24x7x365 asterisk support available?
On Tue, 13 Jan 2004, Jeffrey Paul wrote: Does anyone know of companies or individuals who provide 24x7 asterisk support options? My company does, http://www.internetsolver.com/ dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] threewaycalling ? (Bridge 2 SIP calls?)
On Tue, 13 Jan 2004, WipeOut wrote: If you want it on SIP then you simply have to make sure the phone supports it.. Both my Grandstream and Snom support it so I am sure most phones do.. in fact off the top of my head the only one I know that specifically does not support it is X-Lite.. How are you doing it on the grandstream? I have had no luck getting any three way calling to work. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax on Cisco ATA
What is anyone using the ATA for faxes setting for the LBRCodec, AudioMode, TxCodec, and RxCodec? I am trying to use ulaw/alaw and I have plenty of bandwidth. Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] amaflags question
I am trying to configure cdr on a system. We are using nufone and I have set amaflags=billing on both of their sections in iax.conf. Incoming nufone calls show up in cdr with billing, but outgoing calls still show documentation. What do I need to change? We have a handful of SIP phones, 1 X100P outside line for local, and the rest is via nufone. I don't want inter-system calls to be marked for billing and I don't want local calls to be marked for billing. Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Mon, 8 Dec 2003, John Todd wrote: - a high-density T1 termination system that can handle 8 T1's in a very small amount of rackspace. DS3 de-muxing onboard would be optimal, since anyone with 8 T1's is probably getting a DS3 delivery method, and removing the M13 mux from the rack would be great. Optimally, a 1u rackmount with T3/E3 coax _and_ 28 RJ-45 connections (only 17 of which would be used for E3/E1 muxing) Out of this unit would come IAX2 or (sub-optimally) TDMoE packets to Asterisk peer(s). This solution quickly gets into the discussion of why you might need SS7 for large installations, but I will not address that here, and we'll assume this is all PRI delivery. One thing I would like to see if gr303 in and outbound. I tried to find out from the openss7 people what shape their gr303 stack was in but got no reply. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P echo problems - seem to be fixed now
On Mon, 8 Dec 2003, Jason A. Pattie wrote: Brian West wrote: | Also i'm an SBC victim also.. I feel sorry for you SBC is evil. What about McLeod USA? They aren't necessarily evil, just incompetent. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones!
On Mon, 24 Nov 2003, Nick Bachmann wrote: Ariel Batista wrote: I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 Grandstream told me to go to Ovislink (http://www.ovislink.com/newovislink/IPPhones.asp) to get phones. They're one of the only US places I've been able to find BT-102s. I've got plenty of the 101's and 102's in stock ready to ship. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax over SIP alaw/ulaw
Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] D Channel Bonding
On Wed, 12 Nov 2003, Ray Burkholder wrote: Are the Digium T1/E1 cards capable of D channel bonding for PRI? As in one D channel can service two more PRI lines? NFAS? Not that I know of. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Incoming SIP call
On Thu, 6 Nov 2003, Lal, Deepak (Contractor) wrote: When I get a SIP call, I get the following error: WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 514777, 1) == Spawn extension (incoming, 514777, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten = 514777,1,Dial,Zap/2|10 Is this copied and pasted or retyped? I've seen this before when you have a space in the extension line. There shouldn't be any. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Apple implementation
On Wed, 5 Nov 2003, Charles Hatchette wrote: I am new to Asterisk and Digium card implementation issues. My VAR is strongly recommending using Apple hardware and Yellow Dog Linux for my telephony project, because of his familiarity with this OS. Is the PowerPC an appropriate and stable hardware platform for Digium/Asterisk development? I don't know if the drivers for the cards have been tested on ppc machines. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
On Tue, 4 Nov 2003, Stephen R. Besch wrote: 5) Attempt to balance the hybrid at the 2-line to 4 line interface. Why: 99% of the time, this is where the echo originates and this is where is should be fixed. Unfortunately, this is not for the faint of heart, but if your line card has a hybrid balance adjustment (many don't), use it. Also, with multiple simultaneous calls, this may be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this myself, but you might try something as simple as a 500 ohm variable resistor in series with the ring line and adjust for minimum echo. If it gets worse, you haven't lost anything, just take the resistor out of the line. If it works, measure the value of the resistor when set for minimum echo and replace it with a fixed value resistor. I've had to do similar things to lower loop current. Be sure you get at least 1 watt resistors and put the same size on the tip and ring. Putting resistance on only one wire will throw other things off. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail servermail and fromstring
On Mon, 3 Nov 2003, Senad Jordanovic wrote: The voicemails servermail and fromstring variables should change default values when email voicemail notification gets received by user. I change it, but received mail still shows Asterisk PBX in place of fromstring. Anyone knows is there anything else needs changing? Did you reload after you made the change? dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP NAT
Should it work to have a multi-homed asterisk server with grandstream phones on the internal network and another grandstream phone on the internet and be able to call between them? I set the bindaddr to the external IP and pointed the internal and external grandstream phones to that address. The signalling works fine to call between phones, but when you pick up the ringing phone you get a reorder tone. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, Michael T Farnworth wrote: It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. Second that. It does look/sound cheap. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Clients not connecting
On Wed, 15 Oct 2003, M.A. Ali wrote: I am kind of new to asterisk. Here is a little prolem that I am facing. Here is my problem and questions: I am just adding two gnophone users to my dialplan, all three systems are within lan. 1. in iax.conf: [mako] type=friend auth=pliantext Was this copied and pasted or mistyped? secret=myown context=default host=dynamic permit=0.0.0.0/0.0.0.0 dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream wallmount??
Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialling problems
On Tue, 7 Oct 2003, Brad Waite wrote:
I'm having problems reliably dialling out my FXO card. About 30% of the time
I'll get a your call cannot be completed as dialed. I'm thinking it might be
the dialling speed, but I can't find any configs that change that setting.
We had the same problem and had to modify our extensions like so:
exten = _9NXX,5,Dial(Zap/g1/w${EXTEN:1}||Tr)
Add a w before the number and it will pause a bit.
--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gradual and silent
encroachments of those in power than by violent
and sudden usurpations.- James Madison
___
Asterisk-Users mailing list
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Re: [Asterisk-Users] Transfer from IAX call
On Fri, 3 Oct 2003, Richard Lyman wrote: you'll find that the context is being overwritten. look in chan_iax.c line 1628 and chan_iax2.c line 1645 (or within 3 lines of each) there is a sprintf that is stuff the context, if you comment those out, it should work again. Disclaimer: i have NO CLUE what else this BREAKS!!! THat does look like a problem. Kram - what's the verdict. Is this a bug or a feature? What will changing this line affect? dave Dave Weis wrote: I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called [EMAIL PROTECTED]/1515480 -- Call accepted by 65.127.126.42 (format GSM) -- Format for call is GSM -- IAX2[NuFone]/3 is ringing -- IAX2[NuFone]/3 stopped sounds -- IAX2[NuFone]/3 answered Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Playing 'pbx-transfer' -- Unable to find extension '1' in context 'NANPA' -- Playing 'pbx-invalid' -- Stopped music on hold on Zap/1-1 How do I make this work? dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP problems fixed?
Can anyone confirm that the SIP updates in CVS have fixed the channel leakage and the codec negotiation problem that was happening a few days ago? Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel M Series phones support
On Mon, 29 Sep 2003, Brad Bergman wrote: The M phones from Nortel are digital phones as used with Norstar or Meridian 1 systems. Actually some if not all M8XX and M9XX phones, the 9516 for example (which are now sold by Aastra) are just analog phones with lots of buttons and stuff... they do work nicely with asterisk. Some company has made a card to use meridian phones with the 3com nbx. I found their info - http://www.citel.com/index/index.asp For the price of an nbx on ebay, you could get one and offload to the asterisk system. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Core dumps from Asterisk
On Tue, 30 Sep 2003, Mark Spencer wrote: We should be prtty oeto a 0.5.1 in the net few days. Once Mark puts down the bottle, that is :-) On Wed, 1 Oct 2003, duncan wrote: he mentioned he was using the asterisk 0.5.0 download though. surely this means we should update the 0.5.0 release to solve these problems? can you confirm that it was the asterisk-0.5.0.tar.gz file install that caused these segfaults? It was, I just downloaded the tar file, I haven't touched CVS. I assume the tar file is intended to be the latest stable version, unlike a CVS checkout which would give me a development version? yes thats what i thought was the idea as well. looks like there should be an asterisk-0.5.1.tar.gz release sometime soon then. anyone know of a good checkout day with not much missing or broken? duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: SIP i.e. Is something broken?
On Mon, 29 Sep 2003, Brian Capouch wrote: Christopher J. Wolff wrote: Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Just FYI: I had similar problems for a while, and then I completely scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). That solved the problem. I checkout rebuilt from CVS last night about 10:00 pm, and filed a bug report at that time. It isn't *all* SIP calls for me, btw; my ATA186 works just fine, but my Budgetone won't work with the broken code. . . It's as described in other mails--I can receive calls on the Budgetone but when I make them the RTP part is broken and the calls cut off the second they're set up. I'm seeing the same thing with my budgetones and today's cvs. They are all on the same network and worked on previous versions. I can call voicemailmain and the console says that it is playing but I hear no sound. Then it hangs up automatically. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATM support?
Is there any interest in having ATM support for the various digium T1 cards? dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATM support?
On Fri, 26 Sep 2003, Ryan Butler wrote: On Fri, 2003-09-26 at 09:16, Dave Weis wrote: Is there any interest in having ATM support for the various digium T1 cards? If you mean ATM as well as IMA muxing of ATM T1's (IMA 1.1 please), and the ability for having atm interfaces and pri's on a quad t1 card, then yes :) Do you use pvc's or svc's for voice? Where can I get the correct docs to do this? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Purchasing Grandstream Phones
On Thu, 25 Sep 2003, Aaron Martin wrote: Does anyone know of any reliable supplier for Grandstream phones? Now, I am pretty sure that I haven't done anything to offend these people, and I am pretty sure that Grandstream are not against the idea of selling product.. Has anyone had a Grandstream supplier that actually provides the product!! My company is an authorized Grandstream reseller and I can get you phones wherever you want them in the world :-) dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
On Mon, 15 Sep 2003, Steve Haehnichen wrote: -= On Mon, 15 Sep 2003 15:30:43 -0600, John Brown [EMAIL PROTECTED] said: they require you to purchase 2 phones, No, they just cost more individually: $79.99 today. Versus two for $129.99. Domestic shipping was cheap. and they don't carry BT-102 or AT-286 :) True, that. And they're off doing their own thing, not hip to Asterisk. I'm glad to see another inexpensive source for the Grandstream line. Now if only they would ship the black and blue units.. :) I'll have the black ones available at approximately the end of this month, barring any shipping troubles. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Request for comments on queue statistics
On Tue, 9 Sep 2003, Paul Crick wrote: I have done some trival work with matrix orbital lcd to show some stats counts, calls parked etc Just find lcd a bit small do you have lead on bigger LED signs that you have used b4 ?? I've used a Beta-Brite sign which is pretty similar to a ProLite in functionality, just made by a different company. They're on eBay all the time, search for LED sign as well as the two brand names and you're bound to find something. If you need something bigger try www.translux.com dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] live monitoring
On Mon, 8 Sep 2003, David C. Troy wrote: Just set up an extension that calls it: Exten = 7000,1,ZapBarge Then enter the channel number and pound, and you're good to go. It enters the conference with ZT_CONF_MONITORBOTH option, which as far as I can tell is a listen-only mode -- so no need to mute your phone. Trick with ZapBarge vs. ZapScan is you need to know what channel you want to listen to, as where ZapScan lets you scan through all of them much easier. Which channel number do you enter? sip show channels and show channels both have alphanumeric strings. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openbsd PF firewall ?
On Tue, 2 Sep 2003, Jon Pounder wrote: At 05:17 PM 9/2/2003 -0400, you wrote: Hello. Trying firewalls out. Anyone had any success with an Openbsd PF firewall ? works for us, seems fairly simple to configure, and tamper resistant since it can run in bridge mode with no externally visible ips, so it is impossible for an attacker to gain access to the machine through its external interface. If you are more familiar with linux, I've got a writeup on how to do the same thing at http://www.sjdjweis.com/linux/bridging/ dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DBSaveTree DBLoadTree
On Sun, 31 Aug 2003, Michiel Betel wrote: The db entries persist on reload, on a restart (or crash...) they are gone... On my installation they persist a reload or a stop/start fine. I use it for call forward and dnd settings. dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: zondag 31 augustus 2003 20:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DBSaveTree DBLoadTree On Sun, 31 Aug 2003, Michiel Betel wrote: Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. Doesn't the db stuff persist on disk during reloads? On the same track, I am also looking at exposing DBput DBget to the manager interface, thus making it easy to st global stuff like nightsettings... I will be doing that if you don't get to it within the next week or so. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non Traditional PSTN Trunking
On Mon, 1 Sep 2003, jim b wrote: I am new to Asterisk and wanted to ask a question concerning PSTN trunking. Is there a way to have DID's sent over IP to a switch? I know if One switch has traditional PSTN like a PRI this can be done, but is there a service provider offering this so I dont have to buy any tradtional PSTN trunking? You can get something like that from nufone.net if you want Michigan numbers or toll free. Otherwise you'll need to find someone local to you with a pri and the right equipment. Technically, it is very possible though. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DBSaveTree DBLoadTree
On Sun, 31 Aug 2003, Michiel Betel wrote: Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. Doesn't the db stuff persist on disk during reloads? On the same track, I am also looking at exposing DBput DBget to the manager interface, thus making it easy to st global stuff like nightsettings... I will be doing that if you don't get to it within the next week or so. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Provisioning CO lines
On Thu, 21 Aug 2003, Mike Ciholas wrote: However, while everyone can sell me POTS lines, when I ask about getting these in some sort of digital muxed interface, I seem to confuse the providers. In one case, I was able to get something called channelized T1 which cost a lot and did not actually include the phone service for any of the channels, that was additional. So the cost to go from POTS lines to something digital was extreme, so much more than I can't understand why anyone would have T1 voice interfaces, yet all the PBXes have this and it seems commonly used. I must be doing this wrong. Any t1 service is generally going to cost more than pots service. Not sure why either, with economy of scale, etc. Who is your local telco? 1. Understanding terminology so I can ask for the right thing. Channelized T1 or an ISDN PRI. 2. Advice on when it is reasonable to go POTS versus something else and what that something else is. Unless you need/want DID and the other features, t1 won't be less expensive for you. It will do more but cost more. 3. Feedback on what others are doing with 10-12 lines in the US that may want to expand to ~20 lines. Most people that go with t1 start breaking even or losing less money at 15-20 lines. 4. Interfacing so many POTS lines to Asterisk. I guess that means an FXO channel bank to T1 card? Kind of stupid to go digital/analog/digital in the last 100 feet. Depending on what kind of system you buy initially, ISDN is more flexible than straight channelized. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for real use?
On Wed, 20 Aug 2003, Mike Ciholas wrote: I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. Hedge your bets, pull two cables, and try asterisk. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal
On Mon, 4 Aug 2003, John Todd wrote: Try this: exten = 4001,1,Dial(SIP/gadams,10,r) I don't know how the syntax you've specified will behave; maybe it will work, but it's not any format I've used. Try the syntax above for your Dial line and see if it results in different actions. I can not get my 7960 to work with a non-numeric userid/username. I have to define it in sip.conf as [101] instead of [cisco1] I change the authentication settings in the SIP menu on the phone and in the sip.conf file but it will not register. The version is POS30202 dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virtual fax on the Asterisk box
On Mon, 7 Jul 2003, Andrea Venturi wrote: i recall t38modem (a soft modem working as an h323 endpoint) http://www.openh323.org/t38.html anyone with some experience to share? I tried it a few months ago with a maxtnt and a couple t1's. It would usually receive one page but die on the beginning of the second. I worked with the developer for a while but he wasn't sure what was causing it. It's probably improved by now, however. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Migration to Asterisk - Running off of MerlinLegend system
On 3 Jul 2003, Steven Critchfield wrote: On Thu, 2003-07-03 at 15:11, Steve Creel wrote: Right now, the voicemail system (and auto-attendant) are connected to the switch by 4 analog lines. Logic says that these are FXS cards in the switch, like any other extension. The switch handles an incoming call and transfers it to the auto-attendant. How would such a call be identified to be dropped in the appropriate context? When the phone switch fails to reach someone at an extension, it transfers them to the voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? It is unlikely that you have 4 lines max to send and receive calls to voicemail and auto attendant. It may be some other technology to route TDM to the PC that does voicemail and auto attendant. If this is true, then you won't be able to use those interfaces. That's normal on the legend. You put the 4 station ports in a hunt group and it does coverage for the other extensions. Various configurations are available from 2-12 ports. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 config?
I changed it to use the extension number for the username and secret and now it's working fine. Very strange, but I'm glad it works. Thanks dave On Sun, 15 Jun 2003, Thomas A. Roberts wrote: Dave, In my (limited) experiance with Asterisk, your header in sip.conf must be identical to the 'username' property. Note the example: [phone1] type=friend username=phone1 secret=phone1 host=dynamic defaultip=192.168.1.28 dtmfmode=inband canreinvite=no -- You may want to add this line for your 7960! As for your SIP0002FD3BA8F7.cnf: # SIP Configuration Generic File # Line 1 appearance line1_name: phone1 # Line 1 Registration Authentication line1_authname: phone1 # Line 1 Registration Password line1_password: phone1 If you would like something other than Phone1 displayed on your 7960's line selection screen, add this entry: line1_shortname: Asterisk Test As my first contribution to the list, I hope this helps! Respectfully, Thomas A. Roberts Seegence Corporation www.seegence.com Quoting Dave Weis [EMAIL PROTECTED]: I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: phone1 # Line 1 Registration Password line1_password: phone1 And sip.conf contains: [phone1] type=friend secret=phone1 host=dynamic defaultip=192.168.1.28 dtmfmode=inband I am trying to call between the 7960 and a Grandstream phone. It would work from Cisco - Grandstream, but not vice versa. While fixing that I made it not work either way. It looks like asterisk is sending a 407 to the phone, the phone is trying to authenticate, not succeeding, and getting a 404. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix support
Check the archives, someone got the openline card working. I had been talking with voicetronix about writing a driver for the openswitch cards but they haven't been interested recently. dave On Wed, 28 May 2003, Daniel ANDRE wrote: Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Output
On Wed, 5 Mar 2003 [EMAIL PROTECTED] wrote: Certainly, Oracle does a bangup job there and it's almost as affordable as MySQL. If it means the difference between getting paid and not getting paid, it's considerably less expensive than mysql. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
