Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
I have not found this to be so. As an end user, I have had excellent support from Digium on TDM400p. They have been more responsive the several times I had to call. Even cross shipping replacement cards (CC, required, of course). Cannot fault their support at all. joe a. On 2/16/2008 at 12:53 AM, [EMAIL PROTECTED] wrote: You are kidding, right ??? A small user that just buys one card won't get a good support from Digium. It'll be just a waste of time on the phone. Practically any manufacturer gives similar support including ssh'ing in the users box. Right now they push the user to buy a 4 channel echo canceller which you can get from Octasic for $40. The card with 4 ports is retail around $640. You can get OpenVox or another brand TDM400P compatible for 1/3 of that + $40 for echo canceller. Now that's a Digium high marigin right there .. someone has to pay the CEO salary and the mortgage for a new building :) cheers On 2/15/08, James Finstrom [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. ___ -- 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 ___ -- 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
[asterisk-users] SIP fails to register
Trying to setup SIP to register with a VOIP provider. I am behind a firewall (IPCOP) with NAT. Getting this, in CLI with SIP debug on. Retransmitting #2 (no NAT) to aa.bbb.ccc.ddd:5060: REGISTER sip:voip-xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.xxx:5060;branch=z9hG4bK727a6144;rport From: sip:[EMAIL PROTECTED];tag=eufhksk To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 I suspect there is something, somewhere, where I can tell it the Contact should in fact be my public IP, not the local IP. Anyone know? Or know what else it might be? I am almost 100% certain my credentials are correct. joe a. ___ --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: [asterisk-users] SIP fails to register
Thanks, I believe that is what I was looking for. joe a. On 12/14/2007 at 3:44 PM, Zaheer K. Master [EMAIL PROTECTED] wrote: Hi Joe, In your SIP.conf, under [general] try setting externip=XXX.XXX.XXX.XXX to your public IP address. Hope this helps, Zaheer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Acquisto Sent: Friday, December 14, 2007 2:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP fails to register Trying to setup SIP to register with a VOIP provider. I am behind a firewall (IPCOP) with NAT. Getting this, in CLI with SIP debug on. Retransmitting #2 (no NAT) to aa.bbb.ccc.ddd:5060: REGISTER sip:voip-xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.xxx:5060;branch=z9hG4bK727a6144;rport From: sip:[EMAIL PROTECTED];tag=eufhksk To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 I suspect there is something, somewhere, where I can tell it the Contact should in fact be my public IP, not the local IP. Anyone know? Or know what else it might be? I am almost 100% certain my credentials are correct. joe a. ___ --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 ___ --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 ___ --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: [asterisk-users] Iax and ZAP
On 12/11/2007 at 7:22 AM, Joe Acquisto wrote: I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to register, it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config problem with my setup, but I am currently at a loss. I've found no examples (that make sense) via google. On incoming calls, I get a no such context/extension. I do not have trunk defined for IAX. Outgoing did not work when I did, either. joe a. Managed to get calls to answer (no more no such context/extension), but they just hangup immediately. Below is a sanitized snippet of the debug output: -- tel1*CLI -- Accepting AUTHENTICATED call from nn.nnn.nnn.nnn requested format = ulaw, requested prefs = (g729|ulaw|g726|gsm), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [EMAIL PROTECTED]:1] Answer(IAX2/iaxprovider-3, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(IAX2/iaxprovider-3, SIP/200|20) in new stack Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00070ms SCall: 3 DCall: 00247 [nn.nnn.nnn.nnn:4569] FORMAT : 4 Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 00073ms SCall: 3 DCall: 00247 [nn.nnn.nnn.nnn:4569] == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Goto(IAX2/iaxprovider-3, aa_menu3|s|2) in new stack -- Goto (aa_menu3,s,2) -- Hungup 'IAX2/iaxprovider-3' Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: HANGUP Timestamp: 00084ms SCall: 3 DCall: 00247 [nn.nnn.nnn.nnn:4569] CAUSE CODE : 3 -- A snippet of the dial plan. -- [incoming-iaxprovider] exten = xx,1,Answer() exten = xx,2,Dial(SIP/200,20) exten = xx,3,goto(aa_menu3,s,2) exten = s,n, Hangup - joe a. ___ --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
[asterisk-users] Fwd: re: Iax and ZAP
On 12/12/2007 at 8:35 AM, Joe Acquisto wrote: I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. . . . . . . the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. After correcting my errors, I can receive calls. Outgoing calls remain non functional, insisting on going out over the ZAP channel. I think I would like to maintain the ZAP channels and use iax on occasion. How about something like pressing *99 (or something), presenting a dial tone (or some noise) that would signal to enter the intended number? I am not getting how to direct which channel/trunk to use. joe a. ___ --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
[asterisk-users] Iax and ZAP
I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to register, it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config problem with my setup, but I am currently at a loss. I've found no examples (that make sense) via google. On incoming calls, I get a no such context/extension. I do not have trunk defined for IAX. Outgoing did not work when I did, either. joe a. ___ --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: [asterisk-users] Polycom 601 stops ringing
On 12/7/2007 at 2:33 PM, Doug [EMAIL PROTECTED] wrote: At 10:58 12/7/2007, Joe Acquisto wrote: I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? Is it behind a firewall? joe a. My entire network is behind a firewall, but there is only a switch between asterisk and the phones. joe a. ___ --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
[asterisk-users] Polycom 601 stops ringing
I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? joe a. ___ --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
[asterisk-users] Sip to ATA?
Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding how the phone service works. All I can get is I plug my existing phones and answering machines into the back of the cable modem and am good to go. I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense joe a. ___ --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: [asterisk-users] Sip to ATA?
On 11/27/2007 at 12:26 PM, Ira [EMAIL PROTECTED] wrote: At 06:01 AM 11/27/2007, you wrote: I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense Thanks for both the replies. Hope springs eternal. joe a. ___ --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
[asterisk-users] DST
My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a. ___ --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: [asterisk-users] DST
My thanks to all. Problem resolved with the assistance. joe a. On 11/1/2007 at 1:43 PM, Joe Acquisto [EMAIL PROTECTED] wrote: My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a. ___ --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 ___ --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: [asterisk-users] DST
On 11/1/2007 at 4:22 PM, Turbo Fredriksson [EMAIL PROTECTED] wrote: Quoting Joe Acquisto [EMAIL PROTECTED]: My thanks to all. Problem resolved with the assistance. Would be nice if you posted HOW it was fixed to... I have this exact same problem at home, but the work phones displays time correctly... Sorry, did not want to take up more list space. To quote/snip/paste, from a very recent post (BJ Weschke) (and archives, polycom, etc) -: *** If you've got the files centrally managed, you can update the correct tags in sip.cfg to correct the situation. These are the correct settings for regions affected by the new DST regs: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 *** joe a. ___ --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
[asterisk-users] FAX detection not working
I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs) As I understand it, I must have faxdetect = incoming to enable detection of the fax tone. Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on. In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a FAX call so no one picks up the line. however, it seems the FAX tone is not being detected (I know it is being sent), as the normal ring tone is heard. I must be misunderstanding how this works. Or does not work. joe a. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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: [asterisk-users] FAX detection not working
This can be a partial never mind, I guess. I can see via the CLI that the call is being handled by some FAX related routines. Just not quite the solution I expected. joe a. On 9/29/2007 at 8:56 AM, Joe Acquisto [EMAIL PROTECTED] wrote: I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs) As I understand it, I must have faxdetect = incoming to enable detection of the fax tone. Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on. In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a FAX call so no one picks up the line. however, it seems the FAX tone is not being detected (I know it is being sent), as the normal ring tone is heard. I must be misunderstanding how this works. Or does not work. joe a. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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: [asterisk-users] FAX detection not working
On 9/29/2007 at 3:27 PM, Lee Howard [EMAIL PROTECTED] wrote:
Joe Acquisto wrote:
As I understand it, I must have faxdetect = incoming to enable detection of
the fax tone.
Then, I must have a [fax] context to pickup the line and send it to whatever
extension the FAX device is on.
It's a fax extension in the context where the call is at... not a fax
context in the dialplan.
Lee.
I don't follow. Sorry.
Now might be a good time to post this, since Tzafrir asked, it looks very much
like bits I have seen on the net. I did see what appeared to be the analog_fax
part when checking at CLI.
So, I would surmise it detected the FAX and is trying to deal with it, but the
number derived via LDAPget is hosed? It just ends up hanging up and not
dialing any extension.
{begin snippet]
[ext-fax]
exten = s,1,Answer
exten = s,2,Goto(in_fax|1)
exten = in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax|1)
exten = in_fax,2,Macro(faxreceive)
exten = in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf -
${FAXFILE}.pdf)
exten = in_fax,4,system(mime-construct --to ${EMAILADDR} --subject Fax from
${CALLERID(num)} ${CALLERID(name)} --attachment ${CALLERID(num)}.pdf --type
application/pdf --file ${FAXFILE}.pdf)
exten = in_fax,5,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten = in_fax,6,Hangup
exten = analog_fax,1,GotoIf($[foo${FAX_RX} = foo]?3:2)
exten = analog_fax,2,LDAPget(DIAL=DeviceDial/${FAX_RX})
exten = analog_fax,3,Dial(${DIAL}|20|d)
exten = analog_fax,4,Hangup
exten = out_fax,1,txfax(${TXFAX_NAME}|caller)
exten = out_fax,2,Hangup
exten = h,1,Hangup()
[end snippet]
joe a.
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Re: [asterisk-users] TDM400P (TDM22P) and aux power.
On 9/5/2007 at 10:56 AM, Jason Parker [EMAIL PROTECTED] wrote: Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. joe a. Correct, it is to provide the ringing voltage on the FXS modules. For systems without internal molex connectors available, there is another option. Digium has created an externally powered supply that can be used with these cards. http://www.digium.com/en/products/hardware/analogpwr.php Thanks. I don't know how I missed this when posted, but, better late than never. joe a. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] TDM400P (TDM22P) and aux power.
I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. joe a. ___ --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: [asterisk-users] TDM400P (TDM22P) and aux power.
On 9/5/2007 at 1:06 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Thomas Kenyon wrote: Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Yes, you only need to connect a power supply if you have FXS boards. Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. I don't remember if it is also needed to provide the potential for the line as well, but I cat testify to the fact that you can comfortably run a TDM400P with 4 FXO boards on it and nothing plugged into the PSU header. That is correct. You *only* need the power connector plugged in for FXS modules. FXO modules do not need them. Thanks to all who responded. My hunt for cheap, err, inexpensive, Polycom's continues. joe a. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] Cisco 7960 or 7960G
Is there more than one version of the Cisco 7960? I see some items advertised as 7960 or 7960G, but searching on 7960 only brings up 7960G info, or ambiguous stuff. joe a. ___ --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: [asterisk-users] Cisco 7960 or 7960G
On 9/2/2007 at 9:32 AM, Joe Acquisto [EMAIL PROTECTED] wrote: Is there more than one version of the Cisco 7960? I see some items advertised as 7960 or 7960G, but searching on 7960 only brings up 7960G info, or ambiguous stuff. joe a. A partial never mind, it appears they are two different models. Yet the differences are not readily apparent. joe a. ___ --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: [asterisk-users] OT: DELL Platforms
. . . I bought a nic card for my APC3000 UPS because I was led to believe it could turn on an off all of the 8 power points independently but have never been able to work out how to do this. Anyone know how to do this? Cheers, Dean I have an APC3000 and don't believe that is possible. When I changed the batteries, it looked like the AC outlets were all wired in parallel. Working from memory here. joe a. ___ --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: [asterisk-users] Cisco 7960 sccp
On 9/1/2007 at 7:46 AM, Michiel van Baak [EMAIL PROTECTED] wrote: On 18:59, Fri 31 Aug 07, Joe Acquisto wrote: What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? You'll need the firmware and an TFTP server to get the firmware on the phone. I guess my question is more along the line of how difficult Cisco is about this? I know router firmware is not always just for the asking. Hmm, I guess I *could* ask Cisco . . . joe a. ___ --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
[asterisk-users] Cisco 7960 sccp
What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? joe a. ___ --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
[asterisk-users] Problems with Polycom 300/500/600
Any great disadvantage to using polycom 300/500/600 vs the 301/501/601? joe a. ___ --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
[asterisk-users] FYI
http://www.wired.com/print/politics/security/news/2007/08/wiretap ___ --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: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
On 8/28/2007 at 9:30 AM, John Novack [EMAIL PROTECTED] wrote: Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc 5 year warranty, to name one. Sangoma says their cards will work in ALL modern machines. If they can't make it work ( never seen that ) they will refund. If you have problems, and you give them SSH, they will fix it. John Novack Digium has done this, for me, as well. However, in either case, I have reservations about letting others wack away at my machines, especially if one cannot see what they are doing. No so much not trusting them, but not learning a thing along the way. When I voiced that concern to the Digium techs, they set up a thing called screen (I think it was) to allow me to see and or interact with their session. joe a. ___ --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
[asterisk-users] IAXmodem on Fonality?
Any experiences putting and supporting IAXmodem on Fonality? They themselves do not seem interested. joe a. ___ --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: [asterisk-users] IAXmodem on Fonality?
On 8/26/2007 at 10:41 AM, Patrick [EMAIL PROTECTED] wrote: On Sun, 2007-08-26 at 09:36 -0400, Joe Acquisto wrote: Any experiences putting and supporting IAXmodem on Fonality? They themselves do not seem interested. Do you mean Fonality or Trixbox? If Trixbox you can use the iaxmodem SRPM from http://www.laimbock.com/asterisk/ A few months back when Trixbox still used the SRPMs from laimbock.com there were several people who installed iaxmodem from there too. Regards, Patrick Fonality, not Trixbox. joe a. ___ --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: [asterisk-users] which OS would be fine for asterisk
. . . Personally I recommend SuSE Linux. OpenSuSE without the GUI installed will do just fine. If you want to buy SLES that's fine, but I really don't see the value in it. The value would be live support and access to online updates. Courtesy (for the price) of Novell. There are, of course, some differences between OpenSuse and SLES. I've run Asterisk on SLES 9 and SLES 10 without problems. Your View/Mileage May Vary. joe a. ___ --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: [asterisk-users] which OS would be fine for asterisk
. . . The value would be live support and access to online updates. Courtesy (for the price) of Novell. There are, of course, some differences between OpenSuse and SLES. I've run Asterisk on SLES 9 and SLES 10 without problems. Your View/Mileage May Vary. joe a. With OpenSuSE you get free updates. The support is of no value to me. As stated YMVMV. For some people, the ability to have support and to have updates downloaded and installed automatically, (if desired) might be of value. For others, it would have no value or even a negative value. joe a. ___ --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: [asterisk-users] which OS would be fine for asterisk
. . . Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. Thanks, Steve Totaro I'd have to review the entire thread to see if anyone actually claimed any flavor was best, but can point to the subject that just asked for something fine. For my part, I offered my comments without an axe to grind, no skin in the game. But it certainly might be interesting to see if someone has a best and reasons for it. joe a. ___ --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
[asterisk-users] Setting caller ID on outgoing calls.
Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the outgoing number instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination media is? Meaning, the call would be coming in on a T1, going out on a T1, but ending on a POTS line (which supports caller ID). Thanks for understanding. joe a. ___ --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: [asterisk-users] Setting caller ID on outgoing calls.
I did post recently, under another subject line. But would appreciate some response, as some are telling a client that this is not possible. joe a. On 8/20/2007 at 1:57 PM, Joe acquisto [EMAIL PROTECTED] wrote: Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the outgoing number instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination media is? Meaning, the call would be coming in on a T1, going out on a T1, but ending on a POTS line (which supports caller ID). Thanks for understanding. joe a. ___ --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 ___ --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
[asterisk-users] Forwarding calls, passing Caller ID (or not)
There was a discussion a while back about how to pass Calller ID, when forwarding, as either the calling number, or the forwarding number. Had something to do with scams IIRC, but could not find in browsing the archives. So, is it in the docs? Starting point or full tilt would be appreciated. joe a. ___ --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: [asterisk-users] Major Digium Card Problems
. . . The cost of the wire is not that much, without even shopping around or going through my regular distributor I found this link. http://www.wesbellwireandcable.com/Bare_Tinned_Copper.html?gclid=CNLa25jj6Y0 CFQ1zHgodBychsQ 1000ft = $169. Again, it is not that big of a deal if you plan, explain, and do it correctly. Thanks, Steve The general use of bare copper, IMHO, is not a great idea, especially if having to snake the ground around obstacles, some of which may be conductive. If the idea is a clean, safe, ground. It does have it's place. joe a. ___ --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: [asterisk-users] Blip every 30 seconds?
Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. Perhaps that is what you are hearing? joe a. On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote: Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? ___ --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 ___ --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: [asterisk-users] Royalty for On Hold Music ?
. . . Even if you can find non-original-artist recordings of such music, the *compositions* are registered with BMI and ASCAP, and you'll need blanket licenses to play them. (Well, if you only wanted one or two tracks, you might negotiate specific licenses, but I'm not sure it would be cheaper.) Cheers, -- jra So, if, for instance, someone were to pipe in some broadcast stations, for MOH, that would be a copyright violation? Not that I know how to do that, with *, off the top of my head. joe a ___ --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: [asterisk-users] surge protector?
APC makes a two line unit. PTEL2. But it's two lines in one jack. Another - www.ablecom.com is a bit more Pro Just do a google and take your pick. joe a. On 7/14/2007 at 10:17 PM, Todd H [EMAIL PROTECTED] wrote: I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd ___ --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 ___ --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: [asterisk-users] Call Waiting
On 7/11/2007 at 11:04 AM, Joe acquisto [EMAIL PROTECTED] wrote: Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a hook flash, to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can be done on SIP. What am I not grasping here? About the Analog phone/Asterisk actions. Not too vague, I hope. joe a. OK, so the secret seems to be to flash (press hook button briefly) as normal, the do *0. That takes me to the waiting call. But how to switch back, is still a mystery.Do to various constraints intense testing is not possible at this time. Anyone? joe a. ___ --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
[asterisk-users] Call Waiting
Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a hook flash, to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can be done on SIP. What am I not grasping here? About the Analog phone/Asterisk actions. Not too vague, I hope. joe a. ___ --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: [asterisk-users] Suing Dell||Dull Computers for CID abuse
. . . We let you win, you were terrorists and England's never been good at fighting terrorists. Now you're having the same problem !!! One is stuck by the semi-irony. Those who do not learn from History are doomed to repeat it. However, the current unpleasantness has dis-similar roots. Tho one could say it is the dark heart of Man at the core of it all. . . . Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Steve An Italian explorer called Amerigo Vespucci, I believe. (look it up) joea ___ --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
[asterisk-users] garbled calls
problem - occasional garbled calls, mostly remote users. T1 connected to PSTN, SIP over local LAN and internet to remote users. NAT at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most are Polycom phones, two Aastra's. Start with QoS on LAN switches? No 2x4's please, start with 1x4's. joe a. ___ --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: [asterisk-users] garbled calls
QoS does nothing for you unless you're using MPLS between connections to a degree. (re-stated...) If you're under the impression that you're going to magically place some auto-qos of sorts and your traffic will be magically shaped for high performance, you're semi-mistaken. While it may shape traffic coming internally and how you handle that traffic, unless you have a certain pre-defined service with your provider, most providers won't/don't care how you color your packets. So go ahead and misinterpret QoS by marking DSCP, wasting hours thinking you're going to save .03ms. Fire up Wireshark at the other end and check your classifications and I can bet you lunch for a year that most providers will re-classify or ignore your markings. I was merely stating what exists now and requesting guidance. I am aware that effective QoS requires the co-operation of the ISP (and probably, the end users gear), and do not look forward to messing with any of it. joe a. ___ --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: [asterisk-users] garbled calls
. . . QOS across the internet is pointless and further more doesnt really exist, I would suggest setting qualify=200 in sip.conf so that asterisk will not send a call to the remote end if they are more than 200 milliseconds away. Away, in what sense? Are you referring to packet latency? How does Asterisk measure this? Ping response? joe a. ___ --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: [asterisk-users] garbled calls
. . . Away, in what sense? Are you referring to packet latency? How does Asterisk measure this? Ping response? No, it does NOT measure packet latency. qualify= measures the response time of the remote device to a SIP OPTIONS packet. If the device is busy doing something and does not respond quickly enough the device will be considered unreachable. I suppose I should just go do some reading, but . . . Does this happen only at registration time? Each time a call is attempted? In any case, won't that just kill a call, rather than cure the problem? Isn't this a Dr. Kervorkian solution g? joe a. ___ --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: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Contrary to the opinions of Anglo-Philes, we, here in the Colonies, speak American, not English. In some places, 'Murican. We get to do that, because, back in the late 1700's . . . we won. It is only referred to as English out of a sense of compassion. Oh, so anyway, who was guy Eng you named the country after? joe a. On 7/3/2007 at 6:13 PM, Mark Phillips [EMAIL PROTECTED] wrote: Damn!!! Beat me to it ;-} As an Englishman now living in New Jersey (strangely nowhere near an exit) I have to say that the local idiom and accent leaves a significant amount to be desired. Terms like New Joisey, Shuwa ,wadder, badderies, congradulations etc make me wonder if I'm in an English speaking country at all. I've heard better English spoken in Nigeria. Mark On Tue, 2007-07-03 at 17:07 -0400, Andrew Kohlsmith wrote: On Tuesday 03 July 2007 7:20 am, J. Oquendo wrote: (again) Dell. We know based on someone's accent and lack of proper use of grammar, they are not speaking to us from a location in the USA. How can we validate that such instance is illegal. It You obviously have not been around any city centre in North America if you believe that to be true. :-) -A. ___ --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 ___ --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
[asterisk-users] More FAX over T1
This is a follow up to an earlier post. Looking for a means to individualize incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to enter the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. Currently, there is a dedicated T1 into the Asterisk box. There is a separate bank of 4 POTS lines going into a FAX server. Looking for a way to assign numbers as incoming FAX lines and have them received with the incoming number intact. Having these forwarded to one of the analog numbers is a thought, but I am concerned about various issues, data corruption, etc, going that route. Thoughts vary to second T1, with channel bank, breaking out some DS0's into a channel bank, or finding a T1/fax board (do they exist?), to go directly into the FAX server (PC/linux based) joe a. ___ --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: [asterisk-users] More FAX over T1
. . . With all due respect, this project should be handed over to whomever has authorization to administer the Asterisk box. We can tell you how to do it in Asterisk, but if you can't take our advice, our ability to help you will be severely limited. Thanks. Point taken. I'm, unfortunately, playing a form of monkey in the middle. Seems the Vendor is unwilling to, unable to, or is outrageously priced. I am not privy to any of those discussions. I view this as a learning experience for me. Now, we have many, many fax machines. We have our incoming through PRI, and then redirect to a channel bank. We have no problems with fax reception. When we used a Sangoma card, we did, but now that we're back on Digium hardware, we've been doing well, thus far. Probably had to do with the echo cancellation, but without infinite time to troubleshoot, we just had to get it working. This install uses a Sangoma card. Could you expand on redirect to a channel bank? Could you illuminate the connectivity for me? A single T1 connects to??? Is the Digium card smart as in, can it break out DS0 line(s) on a second port (to go to the channel bank)? I am not that familiar with that technology. As may be evident. joe a. ___ --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: [asterisk-users] More FAX over T1
. . . It looks to me like you have two choices. The first you probably can't do. That is, get a two port board in the Asterisk system with the second T1 going into an Eicon Board in a Hylafax system. Then, you can assign DIDs with whatever web interface you have on this Asterisk system to go to the second T1 port. The second alternative is to get a second T1 and an Eicon card going into a Hylafax Server. This solution has a big monthly expense to it, especially if you aren't fully utilizing all channels on your existing T1. I'm moving towards the first solution being we send out (and receive) large faxes and the IAXModem solution, because of patent issues, is not able to send at the faster speeds. I've received complaints about our slow fax machine. The Eicon card can support the faster transmission. No solution, thus far, seems very cost effective for this client. The Eicon and other T1/fax board are in the 4-7K$ range. As this venture is still in it's infancy, it would not be acceptable to shell out such, at the moment. One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. joe a. ___ --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: [asterisk-users] More FAX over T1
On 6/26/2007 at 3:04 PM, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. That's definitely an idea. Wow, you mean I actually had one??? g If you don't need the Caller ID on the fax (and in most cases, you probably don't), If you mean (not) printed on the FAX, yes (no ?) it is *not* needed. this might be your best solution. Assuming, of course, the faxmodems on Hylafax are picking up the caller ID and you have Caller ID from the phone company. Worthy of investigation. That would take up 2 of your PRI channels, though, per fax reception. That should not be a problem, at this point. joe a. ___ --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
[asterisk-users] FAX over T1
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? joe a. ___ --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
[asterisk-users] OT ? Number portability, land line to Cell
Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. joe a. ___ --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] FXO recommendation
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM: Friday, May 4, 2007, 1:56:09 PM, Joe wrote: Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 6:34 AM: Well this is a digium list, so here will be digium cards recommendation. But You can use a linksys spa3102, that costs about half price of TDM400P. I looked up that linksys device. It does not appear that it can replace ad TDM400P. It is not a card at all but a free standing device. More of an ATA, actaully. Yes it is an ATA with an FXS and an FXO port, and you can use as many as you want instead of one TDM400/TDM800/TDM2400. I don't see how that is possible. This device does not connect to the PCI bus, at all. It has two RJ11 ports that can connect to a LAN, or directly to the asterisk box so it may be possible to make it work, somehow, but it cannot replace a TDM card, which is what I thought you were suggesting. I missed the original posting. Since no one else has spoken up, perhaps I am off base. Please help clear up what I am missing. joe a. ___ --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] OT: Capture Asterisk traffic
. . . man tcpdump indicates that I should be able to use = syntax but it doesn't work as expected. Any further advice appreciated. Cameron When interested in packets, I usually use ethereal and a 4 port hub, plugging the ethereal and asterisk boxs into the hub and uplink the hub to where the asterisk box plugged into. It does require more hardware and a momentary interruption of communications, but seems more flexible and less intrusive (to asterisk) to me. joe a. ___ --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
[asterisk-users] SugarCRM, NO!, Foxpro, SI?
I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet abandoning SugarCRM, and opting to develop their own Visual FoxPro database/CRM? Please don't dump on me now, this is not my idea, I am just asking for comments, to see if my own initial thoughts are reasonably accurate. joe a. ___ --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] SugarCRM, NO!, Foxpro, SI?
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. Thanks. joe a. ___ --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] what version is running?
Try this: # strings -a /usr/sbin/asterisk | grep Digium I get: Asterisk 1.2.16, Copyright (C) 1999 - 2005, Digium, Inc. and others. Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others. but if the version string has been removed from the sources, it's anyones guess... Gordon Indeed, it looks as if the version strings have been removed, But there are core commands at CLI, as Tzafrir mentions. . . joe a. ___ --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
[asterisk-users] what version is running?
A very simple question - what version is running? A CLI - show version does not tell me, only shows info about who compiled, and when. A google and other nefarious devices have not yielded the secret. Sure, I could scour the docs and determine what is features do not exist in all versions, and try them, but there must be a more exact, explict way. joe a. ___ --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] what version is running?
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM: On Sat, 14 Apr 2007, Joe Acquisto wrote: A very simple question - what version is running? A CLI - show version does not tell me, only shows info about who compiled, and when. A google and other nefarious devices have not yielded the secret. Are you sure? I get: dsx*CLI show version Asterisk 1.2.16 built by root @ bob on a i686 running Linux on 2007-03-05 07:25:25 UTC ^^ Gordon Yes, *very* sure. Just did it again, to eliminate brain/time warps (well this time, anyway). Shows what you get, but the version number is missing after Asterisk. Just a few spaces, then the rest. Perhaps the fault lies with those who complied it? joe a. ___ --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] what version is running?
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM: On Sat, Apr 14, 2007 at 07:15:36AM -0400, Joe Acquisto wrote: Gordon Henderson [EMAIL PROTECTED] Wrote: 4/14/2007 7:02 AM: On Sat, 14 Apr 2007, Joe Acquisto wrote: A very simple question - what version is running? A CLI - show version does not tell me, only shows info about who compiled, and when. A google and other nefarious devices have not yielded the secret. Are you sure? I get: dsx*CLI show version Asterisk 1.2.16 built by root @ bob on a i686 running Linux on 2007-03-05 07:25:25 UTC ^^ Gordon Yes, *very* sure. Just did it again, to eliminate brain/time warps (well this time, anyway). Shows what you get, but the version number is missing after Asterisk. Just a few spaces, then the rest. Perhaps the fault lies with those who complied it? SEems like you have a bogus version.h somewhere. this is it, I think: --- tel1:/ # edit /usr/include/asterisk/version.h /* * version.h * Automatically generated */ #define ASTERISK_VERSION #define ASTERISK_VERSION_NUM --- Next: is there any place on the binaries whre the SVN revision is written? I've no idea. Next-best thing would be to sart looking for differences in messages and such from recent commits. I've only this one version to check, at this time. Tzafrir Cohen joe a. ___ --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] what version is running?
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 9:59 AM: On Sat, Apr 14, 2007 at 08:58:57AM -0400, Joe Acquisto wrote: Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/14/2007 7:41 AM: Next-best thing would be to sart looking for differences in messages and such from recent commits. I've only this one version to check, at this time. 1.2 or 1.4? -- Sigh.That is the root question. But, doing a rpm -qa asterisk claims asterisk-1.4.0-1 joe a. ___ --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] Too much silence, perceived delay
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:51 PM: Joe Acquisto wrote: Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM: Have you been able to test this yourself? (Three to four seconds seems inordinately long. That's as bad as a satellite link.) No, not tested by me, I only heard about it today, via email. I don't doubt that they are noticing some delay, I just question how extreme it is. Have you tried tinkering with the gain settings? Adjusting the gain can impact sidetone, which might improve the call experience. No, not yet. Any suggestions as to direction and magnitude? After confirming that they're experiencing what they say they've been experiencing, I would start with the rxgain and increment it by 2 or 3, then test. Applying rxgain of 2 seems to have satisfied the user who was complaining. My own perception of delay finds it acceptable. Could be intermittent, tho, I suppose. joe a. ___ --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] Too much silence, perceived delay
Eric ManxPower Wieling [EMAIL PROTECTED] Wrote: 4/10/2007 3:53 PM: Steve Edwards wrote: On Tue, 10 Apr 2007, Joe Acquisto wrote: My own perception of delay finds it acceptable. Could be intermittent, tho, I suppose. Anytime somebody complains of delay or lag, have them call a cell phone from a cell phone and listen to themselves. Usually their jaw drops :) Then I ask them when was the last time somebody asked them to call back on a land line because the delay was too long. There are different types of delay. There is audio delay and there is dialing delay. I suspect the users are complaining about dialing delays, rather than audio delays. During conversation about this, both types of delay were mentioned. joe a. ___ --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] remote SIP, no audio, or one way audio.
So, a packet trace, at router-internet, was done. Not much to speak of, filtering for phone/* server traffic. While I can see what appears to be a session initiation and they make nice, there appears to be no traffic for audio, at all. Anyone have an example they could share? Or is someone quite well versed in SIP traffic who can read the trace? joe a. Joe Acquisto [EMAIL PROTECTED] Wrote: 4/9/2007 1:42 PM: Hi. Is there a way to isolate what shows on CLI to just the conversation with that extension? There appears to be a lot of stuff unrelated to this extension. Packet traces are not out of the question, but cannot be done today. joe a. Yossi Ben Hagai [EMAIL PROTECTED] Wrote: 4/9/2007 12:56 PM: Hi Joe, The debug trace you've enclosed is a NOTIFY message sent from * for the message waiting feature - and is not related to the call. You can however tell that something is wrong since the message is being retransmitted since the server didn't receive 200 OK in reply - while it could be due to the client being offline or not supporting this feature It could imply a NAT issue so try to recheck your NAT configs. can you post a full trace (starting with the INVITE message)? also you can try to run a sniffer trace on the client side to see if it receives/sends the messages correctly. Joss. On 4/9/07, Joe Acquisto [EMAIL PROTECTED] wrote: I never get this far, apparently. While the connection seems to be made, and calls can be completed (rings, answers) there is no audio. On CLI, I can see what appears to be call being made and connected. These are x-lite phones (for testing, one hopes) there appears to be no codec selection available. I see no CODEC dialog. What I see is six iterations of the below: . . . . --- Retransmitting #6 (NAT) to xx.xx.xx.xx:64909: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport From: nsip:[EMAIL PROTECTED];tag=as67e5c857 To: nsip:[EMAIL PROTECTED];tag=9c58a77e Contact: sip:[EMAIL PROTECTED] Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout Content-Length: 0 - Does this imply anyting to anyone? Call can be made, after this. joe a. ** dave cantera [EMAIL PROTECTED] Wrote: 4/7/2007 3:53 PM: joe, when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer extension' and look for the codec handshaking... make sure both extensions have a compatible codec choice... daveC Using INVITE request as basis request - [EMAIL PROTECTED] Found user '401' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP video format 99 Peer audio RTP is at port 192.168.15.100:5004 *Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format H264 for ID 99 *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - audio=0x2e (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e (gsm|ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.15.100:5004 Peer video RTP is at port 192.168.15.100:5006 Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Joe Acquisto wrote: Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office- - --|sip phone |-ditto Hope that is intelligible. joe a ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] How well does a celldock work with Asterisk?
Steve Prior [EMAIL PROTECTED] Wrote: 4/6/2007 8:30 PM: Steve Prior wrote: I've seen in the wiki that it is possible to use a celldock device to use a cell phone as a PSTN line to Asterisk, but I haven't seen any comments as to how well this actually works. I was thinking about hooking a celldock to a FXO input of my Digium TDM400P card and use it to connect via bluetooth to my RAZR V3C. I am aware of the software solution (chan_bluetooth), but my Asterisk box is a bit far away from where I want to keep the phone so the celldock seems to be the more convenient solution for me. Any comments about the sound quality or issues in making it work? I just found out that the celldock I'm talking about is also called the Dock-N-Talk. I look forward to hearing about experiences in using it with Asterisk. Steve My curiosity is aroused, as well. I would want to use these to allow me to eliminate my POTS lines entirely, and go to cell service. This, due to the poor quality (and comparatively high cost) of Verizon service in this area. Seems I would still need at least one POTS line, for FAX machine. Can't use IAX in any form due to highly asymetrical link at this location. joe a ___ --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] remote SIP, no audio, or one way audio.
I never get this far, apparently. While the connection seems to be made, and calls can be completed (rings, answers) there is no audio. On CLI, I can see what appears to be call being made and connected. These are x-lite phones (for testing, one hopes) there appears to be no codec selection available. I see no CODEC dialog. What I see is six iterations of the below: . . . . --- Retransmitting #6 (NAT) to xx.xx.xx.xx:64909: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport From: nsip:[EMAIL PROTECTED];tag=as67e5c857 To: nsip:[EMAIL PROTECTED];tag=9c58a77e Contact: sip:[EMAIL PROTECTED] Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout Content-Length: 0 - Does this imply anyting to anyone? Call can be made, after this. joe a. ** dave cantera [EMAIL PROTECTED] Wrote: 4/7/2007 3:53 PM: joe, when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer extension' and look for the codec handshaking... make sure both extensions have a compatible codec choice... daveC Using INVITE request as basis request - [EMAIL PROTECTED] Found user '401' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP video format 99 Peer audio RTP is at port 192.168.15.100:5004 *Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format H264 for ID 99 *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - audio=0x2e (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e (gsm|ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.15.100:5004 Peer video RTP is at port 192.168.15.100:5006 Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Joe Acquisto wrote: Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office-- --|sip phone |-ditto Hope that is intelligible. joe a ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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 ___ --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] How well does a celldock work with Asterisk?
. . . I have a Dock-n-Talk at home I use to connect my motorola V60i via a cable so I can't comment on bluetooth. I needed it because for some reason I can only get good cell reception in my bedroom. It works well enough. You can certainly tell you are talking over a cell connection and not a POTS line (it's a little noisier) and now and then I have slight echo. Caller ID is passed through (number only) as well. All in all I'm happy with the purchase. -Dave Can't be worse than my POTS lines. The cable runs here are about 30 years old, and run underground, supposedly, where crossing a government right of way. This run is ancient, as well. Supposedly, during wet weather, this becomes a grounding problem. Certainly the audio quality deteriorates to the point of being almost unusable, during inclement weather and clears up when dry. As I get fair reception, using Nextel, in this area . . . joe a. ___ --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] How well does a celldock work with Asterisk?
. . . Can't be worse than my POTS lines. The cable runs here are about 30 years old, and run underground, supposedly, where crossing a government right of way. This run is ancient, as well. Supposedly, during wet weather, this becomes a grounding problem. Certainly the audio quality deteriorates to the point of being almost unusable, during inclement weather and clears up when dry. The cable sleeve is probably breached and I'll bet the waterproofing jelly has leaked out. If, along with that, you have any cracked insulation on the individual conductors, the results are predictable. This is a common problem on older runs. Sometimes it's just a matter of finding a clean pair in the cable. Have you tried asking Verizon to fix the problem? Don't get me started. That's how I know so much about the situation. They seem disinclined to address the matter, except with happy talk about FIOS in my future. Soon. Right after the metro areas are done. Right. The only fiber around here will be in my diet. joe a. ___ --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] remote SIP, no audio, or one way audio.
Hi. Is there a way to isolate what shows on CLI to just the conversation with that extension? There appears to be a lot of stuff unrelated to this extension. Packet traces are not out of the question, but cannot be done today. joe a. Yossi Ben Hagai [EMAIL PROTECTED] Wrote: 4/9/2007 12:56 PM: Hi Joe, The debug trace you've enclosed is a NOTIFY message sent from * for the message waiting feature - and is not related to the call. You can however tell that something is wrong since the message is being retransmitted since the server didn't receive 200 OK in reply - while it could be due to the client being offline or not supporting this feature It could imply a NAT issue so try to recheck your NAT configs. can you post a full trace (starting with the INVITE message)? also you can try to run a sniffer trace on the client side to see if it receives/sends the messages correctly. Joss. On 4/9/07, Joe Acquisto [EMAIL PROTECTED] wrote: I never get this far, apparently. While the connection seems to be made, and calls can be completed (rings, answers) there is no audio. On CLI, I can see what appears to be call being made and connected. These are x-lite phones (for testing, one hopes) there appears to be no codec selection available. I see no CODEC dialog. What I see is six iterations of the below: . . . . --- Retransmitting #6 (NAT) to xx.xx.xx.xx:64909: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport From: nsip:[EMAIL PROTECTED];tag=as67e5c857 To: nsip:[EMAIL PROTECTED];tag=9c58a77e Contact: sip:[EMAIL PROTECTED] Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout Content-Length: 0 - Does this imply anyting to anyone? Call can be made, after this. joe a. ** dave cantera [EMAIL PROTECTED] Wrote: 4/7/2007 3:53 PM: joe, when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer extension' and look for the codec handshaking... make sure both extensions have a compatible codec choice... daveC Using INVITE request as basis request - [EMAIL PROTECTED] Found user '401' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP video format 99 Peer audio RTP is at port 192.168.15.100:5004 *Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format H264 for ID 99 *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - audio=0x2e (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e (gsm|ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.15.100:5004 Peer video RTP is at port 192.168.15.100:5006 Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Joe Acquisto wrote: Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office-- --|sip phone |-ditto Hope that is intelligible. joe a ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list
Re: [asterisk-users] How well does a celldock work with Asterisk?
Stephen Bosch [EMAIL PROTECTED] Wrote on: 4/9/2007 2:16 PM: Joe Acquisto wrote: Sometimes it's just a matter of finding a clean pair in the cable. Have you tried asking Verizon to fix the problem? Don't get me started. That's how I know so much about the situation. They seem disinclined to address the matter, except with happy talk about FIOS in my future. Soon. Right after the metro areas are done. Right. The only fiber around here will be in my diet. Hehe. Are you in a rural area? -Stephen- Yes. Woods, Hills and Dales, Bears, Deer and that sort of thing. It's an adventure. But not that rural. I am only about 500 ft beyond the max loop length, for DSL, and 1/4 mile from a Time Warner cable run. But, are they flexible? Will they work to make a customer happy? Only in America. (that used to mean something much different). joe a. ___ --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
[asterisk-users] Too much silence, perceived delay
In a system connected to a verizon T1, Digium TE411P (quad T1 echo cancellation), client is complaining it is too quiet. The complaint regards calls over the T1, not in house SIP only calls. Their description indicates they want some earpiece feedback of themselves speaking. Also, they complain that it takes several seconds (3-4) for the other party to respond. That is kind of subjective, I guess. Suggestions? joe a. ___ --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] Too much silence, perceived delay
Stephen Bosch [EMAIL PROTECTED] Wrote: 4/9/2007 7:12 PM: Joe Acquisto wrote: In a system connected to a verizon T1, Digium TE411P (quad T1 echo cancellation), client is complaining it is too quiet. The complaint regards calls over the T1, not in house SIP only calls. Their description indicates they want some earpiece feedback of themselves speaking. Also, they complain that it takes several seconds (3-4) for the other party to respond. That is kind of subjective, I guess. Have you been able to test this yourself? (Three to four seconds seems inordinately long. That's as bad as a satellite link.) No, not tested by me, I only heard about it today, via email. Have you tried tinkering with the gain settings? Adjusting the gain can impact sidetone, which might improve the call experience. No, not yet. Any suggestions as to direction and magnitude? Are you seeing errors on the T1? I'll look, when on site. joe a. ___ --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] FAX thru TDM400p
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:53 PM: On Fri, Apr 06, 2007 at 12:37:47PM -0700, Lee Howard wrote: It's usually built and left in the zaptel source directory where you extracted and built zaptel. If it doesn't get built for you from zttest.c then check the Makefile that it has zttest in BINS like this from mine: BINS=ztcfg torisatool makefw ztmonitor ztspeed zttest fxotune A simpler method: make -C /path/to/zaptel/source zttest which will generate zttest for you. You can use zttest from whatever version of Zaptel you might have. Repeated warnings about clock skew detected. joe a. ___ --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] FAX thru TDM400p
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM: On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote: zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. Do you actually have a timing source? head -c 0 /dev/zap/pseudo Do you get input from there in a resonable time? time head -c 8192 /dev/zap/pseudo Only returns to prompt for each of those. No output to screen. joe a. ___ --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] FAX thru TDM400p
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/7/2007 8:24 AM: On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote: Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM: On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote: zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. Do you actually have a timing source? head -c 0 /dev/zap/pseudo Do you get input from there in a resonable time? time head -c 8192 /dev/zap/pseudo Only returns to prompt for each of those. No output to screen. The second one must have given some output. If the first one gives no errors, you probably have at least a semi-functioning timing source. Yes, sorry, left out time, output follows: foo:~ # time head -c 8192 /dev/zap/pseudo real0m1.029s user0m0.000s sys 0m0.004s foo:~ # joe a ___ --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] FAX thru TDM400p
. . . So your timing source is basically working. -- Tzafrir Cohen And this means . . . any FAX-ing issues must be due to other problems? joe a. ___ --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] Analog phones, dial out
No need, fixed. It was the dialplan. I had commented out some stuff, for other troubleshooting and forgot about it. Once I posted and got some replies and reviewed the basics, there it was. Thanks, all, for the push. joe a. Gustavo Cordeiro [EMAIL PROTECTED] Wrote: 4/5/2007 5:47 PM: Paste here the rules of your extensions.conf for outgoing calls. Sds, Gustavo From: Joe Acquisto [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Analog phones, dial out Date: Thu, 05 Apr 2007 16:28:44 -0400 I have a system with a TDM400p 2FXO, 2FXS. Analog phones work fine, on incoming calls, ring, answer, talk, hangup. However, not so good dialing out. Pickup handset, get dail tone. Cli shows Starting simple switch on Zap/1-1. Press a key and get Hungup Zap/1-1 and get the doot-doot-doot sound. Well, something like that, anyway. It may take a large 2x4, today. joe a. ___ --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 _ Verificador de Segurança do Windows Live OneCare: combata já vírus e outras ameaças! http://onecare.live.com/site/pt-br/default.htm ___ --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 ___ --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
[asterisk-users] FAX thru TDM400p
There seem to have been many discussions about this, so sorry if this is boring. Can one connect a standard fax machine (or fax modem) to an analog port on a TDM400p (as if it were an analog phone, say) and expect it to work reliably? For sending, that is. Detecting and routing the call is another subject (for me). Seems it should, but does not. At least not for me. joe a. ___ --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] FAX thru TDM400p
AFAIK, the FAX targets are normal FAX machines, on the PSTN. What happens is, there appears to be a dial out, and a FAX negotiation, but it always fails. joe a. Bryan M. Johns [EMAIL PROTECTED] Wrote: 4/6/2007 8:49 AM: Is your carrier delivering service via a TDM circuit? It has been our experience that you will get far more reliable fax performance via the method you describe (analog device terminated to a port on a FXS line card) than attempting to use an ATA on the LAN. However, if your carrier is a SIP or IAX trunking provider, your reliability concerns are on the other side of your SIP switch. Bryan Johns Partner Shelton | Johns 1805 Old Alabama Road Suite 200 Roswell, GA 30076 USA Office: 678.248.2637 FindMe: 678.229.1809 Email: [EMAIL PROTECTED] On Apr 6, 2007, at 8:39 AM, Joe Acquisto wrote: There seem to have been many discussions about this, so sorry if this is boring. Can one connect a standard fax machine (or fax modem) to an analog port on a TDM400p (as if it were an analog phone, say) and expect it to work reliably? For sending, that is. Detecting and routing the call is another subject (for me). Seems it should, but does not. At least not for me. joe a. ___ --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 ___ --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
[asterisk-users] Poor analog line quality, wireless base station, FAX-ing
While pondering several issues, poor quality PSTN POTS lines, potential cost savings with multiple cell numbers, the FAX problems over TDM400p, etc, I wondered about: Cell phone Base stations to replace POTS lines. Devices to cradle cell phones and connect to TDM400p, for instance, to mimic PSTN. Are there such beasts, how do they play with asterisk? Will FAX work over such connectivity? joe a. ___ --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] FAX thru TDM400p
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/6/2007 9:51 AM: On Fri, 6 Apr 2007, Joe Acquisto wrote: AFAIK, the FAX targets are normal FAX machines, on the PSTN. What happens is, there appears to be a dial out, and a FAX negotiation, but it always fails. I think what Bryan is asking: Where is your FAX source? If the sending system is coming in on another port on the TDM card then fine, if it's coming in via the Internet (SIP, IAX) then it's going to be variable at best... Gordon Ah. The FAX source in on the same TDM card. joe a. ___ --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] FAX thru TDM400p
Lee Howard [EMAIL PROTECTED] Wrote: 4/6/2007 2:15 PM: Joe Acquisto wrote: AFAIK, the FAX targets are normal FAX machines, on the PSTN. What happens is, there appears to be a dial out, and a FAX negotiation, but What it always fails. What does zttest say about your zap card configuration/installation? If it's not always 99.98% or better then it's due to hardware resource constriction and you need to escalate the zaptel card's priority on the hardware (like putting it at a lower IRQ). Lee. zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. joe a. ___ --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] FAX thru TDM400p
I find the source. IIRC, the problem was there was some dependency (terminology ?), or package that was missing and I got complaints when trying to compile. I can't recall what it was, but it is something that is included in most distro's but not SUSE by default. Is there a way to compile this without screwing up other, existing things? I compile real good, as long as I have a cookbook. joe a. ___ --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] remote SIP, no audio, or one way audio.
J. Oquendo [EMAIL PROTECTED] Wrote: 4/5/2007 6:47 AM: Joe Acquisto wrote: Thanks. And this might go where, in rc.d/rc.firewall.local ? But I don't get it. Isn't this redundant? Since I have port forwarding already. . .? joe a. ___ --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 What this is doing is allowing unfettered access between your PBX and phones. Too many people forget that a VoIP transaction consists of more than just opening up ports 5060 and 5061. This are used for registration/administration, etc., in the case of one way audio, or audio for any matter, this is carried out by RTP on separate ports which will never be the same port unless you have it specified. Summarized: NAT + VoIP = nightmare If at all doable, segment your phones out to a DMZ with VLANs, constructive routing, and ACL's to avoid leveraged security incidents via those phones being opened. Thanks. Do you have recommended switches, capable of supporting VLAN's in an appropriate manner? The cheaper the better, at this point. I have attempted VLAN's several times, for this purpose specifically, using Nortel Baystack 450-24's. Not working as one would expect. Some say these simply do not do VLAN's properly This can go off list, if it is OT. joe a. ___ --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] remote SIP, no audio, or one way audio.
Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office|sip phone |-ditto Hope that is intelligible. joe a ___ --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
[asterisk-users] Analog phones, dial out
I have a system with a TDM400p 2FXO, 2FXS. Analog phones work fine, on incoming calls, ring, answer, talk, hangup. However, not so good dialing out. Pickup handset, get dail tone. Cli shows Starting simple switch on Zap/1-1. Press a key and get Hungup Zap/1-1 and get the doot-doot-doot sound. Well, something like that, anyway. It may take a large 2x4, today. joe a. ___ --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] stun
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/4/2007 3:32 AM: On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon now for the next DA question, where to find it (one)? Google has not been my friend. An alleged spot on sourceforge turned up blank. joe a. +++ www.j4computers.com 845-687-4563 Stone Ridge, NY 12484 +++ ___ --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
[asterisk-users] remote SIP, no audio, or one way audio.
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. ___ --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
[asterisk-users] SIP - choppy sound on local LAN to T1
New install, Asterisk, obviously, Baystack 450 swtiches, verizon T1, Digium 4 port T1 card Some (few) users have had complaints from their clients that sound quality is poor. I do not know if the calls were placed via asterisk, or received via asterisk. If it matters. I believe this is a QoS issue, for the swtiches/infrastructure, wondering what can be done, if any one is familiar with these switches. If they are dog for Voip, recommendations? joe a. ___ --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] stun
. . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? All I should need is the linux daemon thing-let, correct? joe a. ___ --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] remote SIP, no audio, or one way audio.
Easiest method in a nutshell... iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Sorry, this is intended to do what for me? I cannot find -j in the iptables man page. joe a. ___ --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] remote SIP, no audio, or one way audio.
Joe Acquisto [EMAIL PROTECTED] Wrote: 4/4/2007 4:24 PM: Easiest method in a nutshell... iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Sorry, this is intended to do what for me? I cannot find -j in the iptables man page. joe a. I found the -j but am still unclear what this does for the dropped audio issue? joe a. ___ --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] remote SIP, no audio, or one way audio.
J. Oquendo [EMAIL PROTECTED] Wrote: 4/4/2007 5:58 PM: On Wed, 04 Apr 2007, Joe Acquisto wrote: iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j ACCEPT iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j ACCEPT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j REJECT iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p UDP -j REJECT Dur... that should have been -j ACCEPT. Thanks. And this might go where, in rc.d/rc.firewall.local ? But I don't get it. Isn't this redundant? Since I have port forwarding already. . .? joe a. ___ --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
[asterisk-users] stun
Is it possible to install a stun server on asterisk? joe a. ___ --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
[asterisk-users] Aastra 480 i
Getting no service display on aastra 480i. Sip debug shows an unathorized blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out. joe a. ___ --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] Aastra 480 i
Joe Acquisto [EMAIL PROTECTED] Wrote on: 4/2/2007 3:03 PM: Getting no service display on aastra 480i. Sip debug shows an unathorized blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out. joe a. Have configured aastra.cfg and mac.cfg. phone seems to see them, as evidenced by different messages while booting up, but the end result is the same. The time/date are always Sat, Jan 1, 12:00, despite there being a timeserver parameter and timeserver being enabled. Any experiences with these phones? joe a. ___ --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] Aastra 480 i
Michelle Dupuis [EMAIL PROTECTED] Wrote on: 4/2/2007 3:23 PM: You don't need the cfg files (or a tftp) to boot the phones or register. There are some sample configs lying around, but Aastra's are very poorly documented (and their firmware still has big bugs - so don't modify from default too much). We've setup a number of 480i's and got very frustrated with their standard support answer maybe it will be fixed in the next firmware. Start with the basics - look at the name/password/domain used for login. MD Yeah, fell back to that, after the .cfg files did not seem to do anything positive. I have managed to get it to register, I resorted to using the web interface to configure. First, the time magically (not) came up correct. After setting timeserver via the web. Neither the cfg, nor the phone menu, seemed to take. Then, setting proxy and registrar for line 1, from the web interface, seemed to get it registered. I thought local settings always over rode the others. Seems not. joe a. ___ --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
[asterisk-users] TDM400p reliability
What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a. ___ --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
[asterisk-users] TDM400p, no CLI activity
New install, using TDM400p. wctdm is loaded, asterisk loads. Zaptel and zapata.conf are from a working system, same model board, same module locations. CLI command zap show status shows all OK, zap show channels shows nothing defined. Incoming calls show nothing on CLI, analog handsets have no dial tone and, of course, shown nohting on CLI. joe a. ___ --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
