[Asterisk-Users] Help with AGI script calleridnamelookup.agi
I was trying to install and make the calleridnamelookup.agi work with my installation of asterisk. I am not familiar with perl or agi scripts. I would like to know if anyone has gotten the calleridnamelookup.agi to work with a similar installation: I am running Fedora Core 3 2.6.9-1.681_FC3 I have perl installed: perl-5.8.5-9 perl-DBI-1.40-5 perl-Filter-1.30-6 perl-DBD-MySQL-2.9003-5 My configuration only consists of a Digium FXO Card, Sipura, and a couple of phones. I am running asterisk: Asterisk CVS-HEAD-12/29/04-21:13:06 I have installed the asterisk perl package, which includes the AGI script calleridnamelookup.agi. I made the directory: /var/spool/asterisk/calleridlookups I am calling the agi script in extensions.conf: exten = 6145551212,1,AGI,calleridnamelookup.agi exten = 6145551212,2,Dial(SIP/5810,15,r) exten = 6145551212,3,Hangup I get: -- Launched AGI Script /var/lib/asterisk/agi-bin/calleridnamelookup.agi -- AGI Script calleridnamelookup.agi completed, returning 0 -- Executing Dial(SIP/5810|15|r) in new stack -- Called 5810 Nothing shows up in /var/spool/asterisk/calleridlookups I am not sure what else I need to do and I am not exactly sure what the outcome is to be? A file containing all callerid, callername info in a file in /var/spool/asterisk/calleridlookups? TIA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone in German
try SIPPS from Ahead (Nero) klaus Adi Linden wrote: I am looking for a German language softphone. Is there such a thing? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FC2 ztcfg - cannot find channel 2
On Sat, 2005-01-01 at 13:49 +1100, Howard Lowndes wrote: When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) If it's anything like Mandrake and 2.6 kernels this is what I've done. In /etc/modprobe.conf remove all lines with references to ztcfg. Then the only time ztcfg runs is from /etc/init/zaptel When ever you remake zaptel the lines will be reinserted. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Damon Estep wrote: Any PC platform is only as stable as the sum of what you run on it, put a single analog interface in a red hat ES on $10,000 worth of hardware and you will have to reboot every 3 days. I'm not seeing these problem with X101P, nor does any of my (not so many) clients. And all that's boils down to is that: a. I'm lucky and b. I've have helped my luck by using only one card per machine, choosing a good MB and making sure the card don't share IRQ with anything. But the point still remains - any software or hardware that needs to be rebooted every 3 days to work is broken and should not be used. Period. If you use such software or hardware you should find out what's wrong and fix it or switch to something else. The reason that the connection between nightly reboots and MS exists is because: a. MS users they can't fix it - it's propritery and b. MS users can't replace it - for a heck load of reasons not interesting to discuss now. Basically what we're saying is that nightly reboot is simply not an option, except in MS shops, that for some reasons are willing to accept such low quality. How and why this comes about is left as excersize to the alert reader... ;-) Gilad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone in German
Adi Linden wrote: I am looking for a German language softphone. Is there such a thing? DIAX has german language support. rgds pos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and Zapatel
Adi Linden wrote: On Thu, 30 Dec 2004, Lyle Giese wrote: Is your X100P set for loop start or Kewl Start? I am betting loop start, try changing to ks instead. Lyle This is what I have in /etc/asterisk/zapata.conf so it should be Kewl Start. It might be that your local telco does not supply disconnect supervision. Try adding: busydetect=yes busycount=6 And if you're lucky Asterisk will guesstimate hangup based on that busy signal you hear when the line is disconnected by the other hand (it takes about 10 seconds with bustcount=6). A word of warning: turning this might cause random disconnect in the middle of calls, so test test test. Gilad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Greg - Cirelle Enterprises wrote: Are you running a stable (v 1.0 - 1.0.3) or cvs Asterisk CVS-v1-0-10/03/04 I've upgraded two months ago to get a feature I wanted (SMS support). It should be round about Asterisk 1.0.2 Gilad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RFC3389 support incomplete
We try a X-Lite client from remote to connect to my * I can call X-Lite and X-Lite can call me. However, X-Lite can hear my voice, while I cannot hear him. add these to sip.conf disallow=all allow=alaw *CLI shows *CLI (date) NOTICE[4637] rtp.c:317 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible RFC3389: 5 bytes, level 4 ... Turn off silence suppression on the client... I tried in my sip.conf to change dtmfmode from inband to info and to rfc2833 without success. dtmf settings has nothing to do with rfc3389 roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX media
In IAX protocol, both rtp and signaling are handled on the same port, so the Asterisk is always in the path of rtp traffic. Am I right? If yes, is there anyway to set Asterisk just as signal proxy ? IAX doesn't use RTP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FC3 compile with new 2.6.10 fails
On Fri, 2004-12-31 at 21:14 -0500, Jerry Geis wrote: All, I have FC3 fedora core 3 and just installed and compiled 2.6.10. after rebooting I attempted to recompile zaptel-1.0.3. I did a make clean then make. I got the following errors. Any suggestions? --- /usr/src/digium/zaptel-1.0.3/torisa.c:1139: warning: 'set_tor_base' defined but not used CC [M] /usr/src/digium/zaptel-1.0.3/wcusb.o CC [M] /usr/src/digium/zaptel-1.0.3/wcfxo.o CC [M] /usr/src/digium/zaptel-1.0.3/wcfxs.o /usr/src/digium/zaptel-1.0.3/wcfxs.c: In function `wcfxs_interrupt': /usr/src/digium/zaptel-1.0.3/wcfxs.c:473: sorry, unimplemented: inlining failed in call to 'wcfxs_proslic_check_hook': function body not available /usr/src/digium/zaptel-1.0.3/wcfxs.c:810: sorry, unimplemented: called from here /usr/src/digium/zaptel-1.0.3/wcfxs.c:474: sorry, unimplemented: inlining failed in call to 'wcfxs_proslic_recheck_sanity': function body not available /usr/src/digium/zaptel-1.0.3/wcfxs.c:812: sorry, unimplemented: called from here /usr/src/digium/zaptel-1.0.3/wcfxs.c:472: sorry, unimplemented: inlining failed in call to 'wcfxs_voicedaa_check_hook': function body not available /usr/src/digium/zaptel-1.0.3/wcfxs.c:814: sorry, unimplemented: called from here make[2]: *** [/usr/src/digium/zaptel-1.0.3/wcfxs.o] Error 1 make[1]: *** [_module_/usr/src/digium/zaptel-1.0.3] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.10' make: *** [linux26] Error 2 What version of gcc? To overcome this with other programs I had to install an older version of gcc and then configure to use that for those compiles. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is wiki drunk
is there any problem with wiki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] final call for Departments
On Thu, Dec 30, 2004 at 11:16:01PM -0800, Alspach Family wrote: I don't want to sound like a TV evangelist from the 80's and 90's but if you have it to give, please do. We have operators standing by to accept your donation. All you have to do is PayPal it to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (Note, this is not me. Rob is the guy doing all the work.) Just dropped in my $10.00, perhaps a few other folks can do likewise. -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is wiki drunk
is there any problem with wiki probably too much champagne last night. I only get a connection closed. roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is wiki drunk
is there any problem with wiki __ It seems to be down thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is wiki drunk
Must be the whole site is down Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed Sent: Friday, December 31, 2004 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] is wiki drunk is there any problem with wiki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Any PC platform is only as stable as the sum of what you run on it, put a single analog interface in a red hat ES on $10,000 worth of hardware and you will have to reboot every 3 days. I'm not seeing these problem with X101P, nor does any of my (not so many) clients. And all that's boils down to is that: a. I'm lucky and b. I've have helped my luck by using only one card per machine, choosing a good MB and making sure the card don't share IRQ with anything. But the point still remains - any software or hardware that needs to be rebooted every 3 days to work is broken and should not be used. Period. If you use such software or hardware you should find out what's wrong and fix it or switch to something else. I think you've really nailed it on the head with the above statement. The problem is that everyone keeps harping on shared interrupts, repeating statements that others have made, etc, but no one is actually making any attempt to identify the root problem. And, primarily because this seems to be a very technical issue with digium cards only, where the majority of developers and those with the skills necessary to identify the root cause don't use digium cards. These systems (motherboards and all) don't have a problem with 99.9% of other vendor's cards and drivers, but yet the * resolution constantly boils down to things like swapping motherboards instead of identifying what the issue really is in terms of certain digium cards. The reason that the connection between nightly reboots and MS exists is because: a. MS users they can't fix it - it's propritery and b. MS users can't replace it - for a heck load of reasons not interesting to discuss now. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is wiki drunk
Working fine right now. (10:25 CST) From: Jeff Glassman [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] is wiki drunk Date: Sat, 1 Jan 2005 11:03:37 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Must be the whole site is down Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed Sent: Friday, December 31, 2004 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] is wiki drunk is there any problem with wiki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dies every hour
Happy new year! The last 3 months my asterisk has run perferct. But after I have set 15 new SNOM 190 phones on it dies every hour. Nothing to se in CLI ore in the log. It dies with exit status 139 Is there anyone who has an idea of what is wrong - ore any tip on how to test. /var/log/asterisk/messages Jan 1 15:18:17 DEBUG[7193]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jan 1 15:18:25 DEBUG[7193]: Auto destroying call '[EMAIL PROTECTED]' Jan 1 15:19:25 DEBUG[7193]: Setting NAT on RTP to 0 Jan 1 15:19:26 DEBUG[7193]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jan 1 15:19:26 DEBUG[7193]: Setting NAT on RTP to 0 Jan 1 15:19:26 DEBUG[7193]: Setting NAT on RTP to 0 Jan 1 15:19:30 VERBOSE[7283]: Asterisk Event Logger Started /var/log/asterisk/event_log Jan 1 15:19:30 VERBOSE[7283]: == Manager registered action Ping Jan 1 15:19:30 VERBOSE[7283]: == Manager registered action Events Jan 1 15:19:30 VERBOSE[7283]: == Manager registered action Logoff Jan 1 15:19:30 VERBOSE[7283]: == Manager registered action Hangup Jan 1 16:18:24 DEBUG[7317]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jan 1 16:18:26 DEBUG[7317]: Setting NAT on RTP to 0 Jan 1 16:18:26 DEBUG[7317]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jan 1 16:18:26 DEBUG[7317]: Setting NAT on RTP to 0 Jan 1 16:18:26 DEBUG[7317]: Setting NAT on RTP to 0 Jan 1 16:18:31 VERBOSE[7407]: Asterisk Event Logger Started /var/log/asterisk/event_log Jan 1 16:18:31 VERBOSE[7407]: == Manager registered action Ping Jan 1 16:18:31 VERBOSE[7407]: == Manager registered action Events Jan 1 16:18:31 VERBOSE[7407]: == Manager registered action Logoff Jan 1 16:18:31 VERBOSE[7407]: == Manager registered action Hangup .. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I'm not seeing these problem with X101P, nor does any of my (not so many) clients. And all that's boils down to is that: a. I'm lucky and b. I've have helped my luck by using only one card per machine, choosing a good MB and making sure the card don't share IRQ with anything. But the point still remains - any software or hardware that needs to be rebooted every 3 days to work is broken and should not be used. Period. If you use such software or hardware you should find out what's wrong and fix it or switch to something else. I think you've really nailed it on the head with the above statement. The problem is that everyone keeps harping on shared interrupts, repeating statements that others have made, etc, but no one is actually making any attempt to identify the root problem. And, primarily because this seems to be a very technical issue with digium cards only, where the majority of developers and those with the skills necessary to identify the root cause don't use digium cards. These systems (motherboards and all) don't have a problem with 99.9% of other vendor's cards and drivers, but yet the * resolution constantly boils down to things like swapping motherboards instead of identifying what the issue really is in terms of certain digium cards. I have never once had an issue with a card from Digium. The real issue is some people seem to think it's OK to use a 20 dollar motherboard and shove a T1 card in the thing and expect it to play nice. Come on guys I see this all the time... Not all hardware is good hardware that plays nice with other hardware. bkw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RFC3389 support incomplete
I'm not sure but I think it's not about DMTF, but about the silence supression or VAD (voice activity detection) that * doesn't support. Try unabling it in the client. We try a X-Lite client from remote to connect to my * I can call X-Lite and X-Lite can call me. However, X-Lite can hear my voice, while I cannot hear him. *CLI shows *CLI (date) NOTICE[4637] rtp.c:317 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible RFC3389: 5 bytes, level 4 ... I tried in my sip.conf to change dtmfmode from inband to info and to rfc2833 without success. Can anybody give me a hint? bye Ronald Miguel Ruiz Velasco __ Do you Yahoo!? Send holiday email and support a worthy cause. Do good. http://celebrity.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TE410P not Interrupting
In article [EMAIL PROTECTED], Eric Bishop [EMAIL PROTECTED] wrote: Hi all, Just got a brand new server and a Digium TE410P. I get the sequential (knight rider) lights before loading the zaptel driver. As soon as I load the driver all loghts go off. It appears the card is not generating interrupts. [EMAIL PROTECTED] ~]# cat /proc/interrupts CPU0 0: 111005341IO-APIC-edge timer 1: 9IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 66IO-APIC-edge i8042 14: 7870IO-APIC-edge ide0 185: 0 IO-APIC-level t4xxp 193: 26141 IO-APIC-level cciss0 201:1139611 IO-APIC-level eth0 NMI: 0 LOC: 111010062 ERR: 0 MIS: 0 [EMAIL PROTECTED] ~]# Have also tried replacing the card, changinf PCI slots and messing with the BIOS all with the same result. If anyone can help I would be very grateful.. I have found this on one particular server (a 1U industrial server), and it was traced to a faulty backplane not connecting the IRQ line. As a test, try replacing the TE410P with a different PCI card that you know works in another computer, e.g. an Ethernet card. See if it initializes and generates interrupts. If not, then your problem is on the PCI bus somewhere. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I'm not seeing these problem with X101P, nor does any of my (not so many) clients. And all that's boils down to is that: a. I'm lucky and b. I've have helped my luck by using only one card per machine, choosing a good MB and making sure the card don't share IRQ with anything. But the point still remains - any software or hardware that needs to be rebooted every 3 days to work is broken and should not be used. Period. If you use such software or hardware you should find out what's wrong and fix it or switch to something else. I think you've really nailed it on the head with the above statement. The problem is that everyone keeps harping on shared interrupts, repeating statements that others have made, etc, but no one is actually making any attempt to identify the root problem. And, primarily because this seems to be a very technical issue with digium cards only, where the majority of developers and those with the skills necessary to identify the root cause don't use digium cards. These systems (motherboards and all) don't have a problem with 99.9% of other vendor's cards and drivers, but yet the * resolution constantly boils down to things like swapping motherboards instead of identifying what the issue really is in terms of certain digium cards. I have never once had an issue with a card from Digium. The real issue is some people seem to think it's OK to use a 20 dollar motherboard and shove a T1 card in the thing and expect it to play nice. Come on guys I see this all the time... Not all hardware is good hardware that plays nice with other hardware. The only issue I have with that is there are several people with digium T1 and TDM cards in their systems, and its always the TDM that goes out to lunch; not the T1. No doubt there are less then desirable mobos around (and probably lots of them), but that doesn't explain why stability of the TDM's very different from a T100P (both with Intel 537 chips). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp app_rxfax - the sending software loops
Hi, After a hiatus of several months, I decided to try spandsp again because it is such an excellent addition to asterisk. Using 1.0.3 and spandsp 0.0.1k. When I send a fax from a laptop software fax that works with every machine I've ever had to fax to, I get something like this as staus return from the fax sending software: - Getting Local ID - Getting Parameters and it loops like this forever until I cancel. On the * end, the fax is detected properly, the app called and it even sends my a bad PDF of about 2k. One of our customers has a regular hardware fax and it can't talk to spandsp either, or couldn't last time I tried. I have gotten spandsp to work with one fax only, that was jfax (now j2.com) Ironically, J2 is exactly the service I want to replace with rxfax! I've seen plenty about frame errors and such but nothing about the symptoms I describe. Anyone have any ideas? Otherwise I'll keep waiting and retrying. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy message on ISDN cards?
Hi all, I'm experiencing some problems with i4l and i can't find a solution. I'm using Eicon Diva 1 BRI Eicon Diva Server 4 Bri A ISDN PBX where I connect the first ISDN card (exten 204 in the ISDN PBX) and a ISDN phone (exten 210 in the ISDN PBX) Suse 8.1 with ISDN4Linux Asterisk 1.0 I'm trying to make phone calls from a softphone (Windows Messenger 4.6) to the ISDN phone but I get this: Asterisk Ready. *CLI -- Executing Dial(SIP/edu-ee6e, Modem/g1/204:210) in new stack Urgent handler Jan 1 18:51:19 WARNING[29364]: chan_modem_i4l.c:601 i4l_dial: Outgoing MSN edu not allowed (see outgoingmsn=,*, in modem.conf) -- Called g1/204:210 Urgent handler -- Modem[i4l]/ttyI0 is busy Urgent handler -- Hungup 'Modem[i4l]/ttyI0' Urgent handler Urgent handler == Everyone is busy/congested at this time == Auto fallthrough, channel 'SIP/edu-ee6e' status is 'CHANUNAVAIL' Urgent handler I see how the modem activates using imon (ISDN monitor) and tries to connect. When I add outgoingmsn=edu in modem.conf the modem does not respond, and I get the following: Asterisk Ready. *CLI -- Executing Dial(SIP/edu-c490, Modem/g1/204:210) in new stack Urgent handler Urgent handler -- Called g1/204:210 (...long time passed...) Urgent handler -- Hungup 'Modem[i4l]/ttyI0' Urgent handler Urgent handler Jan 1 18:58:55 WARNING[29394]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Critical Response) Calls from the ISDN phone to a softphone WORK. Can anyone help me??? Thanks in advance ;) Eduardo. My configuration is the following: ** modem.conf ** [interfaces] context=INCOMING_ISDN driver=i4l language=en type=autodetect dtmfmode=i4l dialtype=tone mode=immediate group=1 msn=204 incomingmsn=*,0,210 outgoingmsn=* device=/dev/ttyI0 ;device=/dev/ttyI1 ;device=/dev/ttyI2 ;device=/dev/ttyI3 ;device=/dev/ttyI4 ;device=/dev/ttyI5 ;device=/dev/ttyI6 ;device=/dev/ttyI7 *** extensions.conf *** [INCOMING-SIP] exten=6000,1,Dial,SIP/edu exten=7000,1,Dial,SIP/edu2 exten=4000,1,Dial,Modem/g1/204:210 [INCOMING_ISDN] exten=s,1,Wait(1) exten=s,2,Answer exten=s,3,Dial,SIP/edu * sip.conf * [edu] type=friend username=edu host=dynamic diallow=all allow=gsm ;allow=ulaw allow=alaw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy message on ISDN cards?
Hi, I had the same problem when i tried i4l, and as far as I remember the solution was to set the outgoing msn to the msn of the isdn-line. From my old modem.conf: incomingmsn=* outgoingmsn=123456,123457 device = /dev/ttyI0 device = /dev/ttyI1 best regards, Nils On Sat, 1 Jan 2005, Eduardo López Martínez wrote: Hi all, I'm experiencing some problems with i4l and i can't find a solution. I'm using Eicon Diva 1 BRI Eicon Diva Server 4 Bri A ISDN PBX where I connect the first ISDN card (exten 204 in the ISDN PBX) and a ISDN phone (exten 210 in the ISDN PBX) Suse 8.1 with ISDN4Linux Asterisk 1.0 I'm trying to make phone calls from a softphone (Windows Messenger 4.6) to the ISDN phone but I get this: Asterisk Ready. *CLI -- Executing Dial(SIP/edu-ee6e, Modem/g1/204:210) in new stack Urgent handler Jan 1 18:51:19 WARNING[29364]: chan_modem_i4l.c:601 i4l_dial: Outgoing MSN edu not allowed (see outgoingmsn=,*, in modem.conf) -- Called g1/204:210 Urgent handler -- Modem[i4l]/ttyI0 is busy Urgent handler -- Hungup 'Modem[i4l]/ttyI0' Urgent handler Urgent handler == Everyone is busy/congested at this time == Auto fallthrough, channel 'SIP/edu-ee6e' status is 'CHANUNAVAIL' Urgent handler I see how the modem activates using imon (ISDN monitor) and tries to connect. When I add outgoingmsn=edu in modem.conf the modem does not respond, and I get the following: Asterisk Ready. *CLI -- Executing Dial(SIP/edu-c490, Modem/g1/204:210) in new stack Urgent handler Urgent handler -- Called g1/204:210 (...long time passed...) Urgent handler -- Hungup 'Modem[i4l]/ttyI0' Urgent handler Urgent handler Jan 1 18:58:55 WARNING[29394]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Critical Response) Calls from the ISDN phone to a softphone WORK. Can anyone help me??? Thanks in advance ;) Eduardo. My configuration is the following: ** modem.conf ** [interfaces] context=INCOMING_ISDN driver=i4l language=en type=autodetect dtmfmode=i4l dialtype=tone mode=immediate group=1 msn=204 incomingmsn=*,0,210 outgoingmsn=* device=/dev/ttyI0 ;device=/dev/ttyI1 ;device=/dev/ttyI2 ;device=/dev/ttyI3 ;device=/dev/ttyI4 ;device=/dev/ttyI5 ;device=/dev/ttyI6 ;device=/dev/ttyI7 *** extensions.conf *** [INCOMING-SIP] exten=6000,1,Dial,SIP/edu exten=7000,1,Dial,SIP/edu2 exten=4000,1,Dial,Modem/g1/204:210 [INCOMING_ISDN] exten=s,1,Wait(1) exten=s,2,Answer exten=s,3,Dial,SIP/edu * sip.conf * [edu] type=friend username=edu host=dynamic diallow=all allow=gsm ;allow=ulaw allow=alaw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Nils Segerdahl --- Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03 http://www.upsys.seFax: (+46) (0)18 56 80 49 --- Jan 1 Anniversary of the Triumph of the Revolution in Cuba Jan 1 Castro expels Cuban President Batista, 1959 Jan 1 Churchill delivers his Iron Curtain speech, 1947 Jan 1 First Rose Bowl; Michigan 49 - Stanford 0, 1902 Jan 1 ATT officially divests its local Bell companies, 1984 Jan 1 The Epoch (Time 0 for UNIX systems, Midnight GMT, 1970) --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Rich Adamson wrote: The only issue I have with that is there are several people with digium T1 and TDM cards in their systems, and its always the TDM that goes out to lunch; not the T1. No doubt there are less then desirable mobos around (and probably lots of them), but that doesn't explain why stability of the TDM's very different from a T100P (both with Intel 537 chips). You know, this is sort of a crazy guess, but I think the power on the telephone line has a lot to do with the flakiness of these cards. I've seen something with the X101P that lead me to think so: I have two cards and two lines. I also own a small UPS that happend to have a jack for a phone line, to act as a power cleaner and I've put the line that goes to one of these cards there. Now, in two different occasions, during a power out the UPS signaled the server which shutdown gracefully and when the power was back on , the card whose line went through the UPS was in a Red Alert state. Only taking out the line from the UPS and putting it directly into the telco socket a couple of time cleared this alaram (no, even rebooting did not help). After these two inceidents I simply kept the line directly to the socket (like the second card) and we had several power outages since (don't ask) and this did not happen. So my guess is - what makes the FXO/FXS more sensative then the PRI cards is the power on the line. Or not. Did I mention it's crazy guess? :-) Gilad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
I've seen something with the X101P that lead me to think so: I have two cards and two lines. I also own a small UPS that happend to have a jack for a phone line, to act as a power cleaner and I've put the line that goes to one of these cards there. Surge arrestors used for POTS lines aren't the same as used for digital circuit. I'm not surprised you had trouble. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FC2 ztcfg - cannot find channel 2
On Sat, 2005-01-01 at 21:05, Dave Cotton wrote: On Sat, 2005-01-01 at 13:49 +1100, Howard Lowndes wrote: When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) If it's anything like Mandrake and 2.6 kernels this is what I've done. In /etc/modprobe.conf remove all lines with references to ztcfg. I have already done that. Then the only time ztcfg runs is from /etc/init/zaptel This is the only place where I have ztcfg run from and I still get the problem. When ever you remake zaptel the lines will be reinserted. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip reload - Hang
I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or 'sip reload' as it hangs and stops processing calls or responding on the CLI. I tracked this down to something dealing with the peers I have in sip.conf. However, I removed all peers and just placed a simple friend in sip.conf (right from the sample file): [xlite1] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid=Jane Smith 5678 host=dynamic nat=yes ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw From the Asterisk CLI, I do: sip show peers It shows the one peer and then sip reload and then sip show peers and it is blank with the system being hung. I've searched high and low for a solution, reinstalled Linux (Fedora Core 3), reinstalled Asterisk (I enabled the ztdummy module), and have used the sample config with the same results (for the sample, I just uncomment the above). Since I used the sample without modifications, the fact that I'm using it behind NAT shouldn't make a difference. Any ideas on what is causing this? Is there any additional information I can provide for assistance? Thanks. -- Scott Gruby mailto:[EMAIL PROTECTED] http://www.gruby.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Qs about FXO/FXS cards
Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to get the TDM400P with the necessary FXO/FXS boards - can I expect the installation to be somewhat straightforward? Any tips to avoid grief? Would it make more sense to get 3 cheap X100P's and use some kind of ATA for the FXS? Will obviously save a whole bunch of money, but will there be significant added complexity? -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
On Sat, 2005-01-01 at 14:06 -0500, James wrote: I've seen something with the X101P that lead me to think so: I have two cards and two lines. I also own a small UPS that happend to have a jack for a phone line, to act as a power cleaner and I've put the line that goes to one of these cards there. Surge arrestors used for POTS lines aren't the same as used for digital circuit. I'm not surprised you had trouble. There was no mention of digital circuits other than the opinion that a digital line would have a more controlled voltage supply than the analog lines. The surge arrestor he used was on a analog line. I have wondered a bit as well as it has been mentioned before about telcos performing some form of test occasionally usually late at night that would trip up various analog equipment. While I doubt the test would account for many of the problems, it illustrates the multitude of potential variables -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip reload - Hang
Scott Gruby wrote: sip show peers and it is blank with the system being hung. snip Any ideas on what is causing this? Is there any additional information I can provide for assistance? You can start with actually telling us what version of Asterisk you are using, and how you installed it (from a binary distribution, from source tarball, from CVS, etc.). We also need to know whether you've added any patches or any other additional modules. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to get the TDM400P with the necessary FXO/FXS boards - can I expect the installation to be somewhat straightforward? Any tips to avoid grief? I think the best that anyone can estimate at this time is that a problem exits of some sort and it might be related to a combination of factors, one of which seems to involve specific motherboard designs. There are far too many people complaining about the same issues with the TDM card, several of which opened cases with digium support and got no response. It should be fairly obvious from the many postings on this list since that card was announced that something is not right, and at least some of those systems have digiums T1 cards on the exact same pci bus that are operating just fine. If you read through the archives, you'll see a number of people flapping their jaws using adjectives and adverbs about what they think the issue happens to be, but the majority (if not all) don't have a TDM card and apparently wouldn't touch one with a ten foot pole. What is obvious at this point is: a. no one on the -user list is going to fix (or even hint with any degree of authority) the root-cause b. don't ever post anything to the -dev list regarding a TDM card as that is NOT the forum for digium cards or drivers, c. digium support is not addressing the issue, and, d. the amount of effort required to support the TDM card (stop *, restart the drivers, start *) in its present condition is far greater then what any reasonable non-technical customer will endure. Since this has been an on-going battle, I'd suggest avoiding the TDM card totally, or, take the 50-50 risk to see how high you can raise your frustration level. Would it make more sense to get 3 cheap X100P's and use some kind of ATA for the FXS? Will obviously save a whole bunch of money, but will there be significant added complexity? Three x100p's are likely to cause even more issues due to the high level of system interrupts. (Note: digium has removed them from their web site.) Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
What exactly are people seeing when they have issues with their TDM card? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Saturday, January 01, 2005 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to get the TDM400P with the necessary FXO/FXS boards - can I expect the installation to be somewhat straightforward? Any tips to avoid grief? I think the best that anyone can estimate at this time is that a problem exits of some sort and it might be related to a combination of factors, one of which seems to involve specific motherboard designs. There are far too many people complaining about the same issues with the TDM card, several of which opened cases with digium support and got no response. It should be fairly obvious from the many postings on this list since that card was announced that something is not right, and at least some of those systems have digiums T1 cards on the exact same pci bus that are operating just fine. If you read through the archives, you'll see a number of people flapping their jaws using adjectives and adverbs about what they think the issue happens to be, but the majority (if not all) don't have a TDM card and apparently wouldn't touch one with a ten foot pole. What is obvious at this point is: a. no one on the -user list is going to fix (or even hint with any degree of authority) the root-cause b. don't ever post anything to the -dev list regarding a TDM card as that is NOT the forum for digium cards or drivers, c. digium support is not addressing the issue, and, d. the amount of effort required to support the TDM card (stop *, restart the drivers, start *) in its present condition is far greater then what any reasonable non-technical customer will endure. Since this has been an on-going battle, I'd suggest avoiding the TDM card totally, or, take the 50-50 risk to see how high you can raise your frustration level. Would it make more sense to get 3 cheap X100P's and use some kind of ATA for the FXS? Will obviously save a whole bunch of money, but will there be significant added complexity? Three x100p's are likely to cause even more issues due to the high level of system interrupts. (Note: digium has removed them from their web site.) Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. I have experienced nothing but grief when trying to set up the PSTN part of the SPA-3000. Everything from crackly audio to fast busies. Has anybody tried the Clipcomm CG-410 4-port FXO gateway (http://www.voipsupply.com/product_info.php?products_id=241)? -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
I personally have seen: 1) power alarms on FXO ports. It appears that rebooting the server is the fix for this problem. I have not seen it enough to give definative answers to questions about unloading/reloading the kernal modules to clear this condition. But my experience is that once a port goes into power alarm, it's basically dead and ignores incoming calls at that point. 2) Two TDM cards, one worked, one didn't in same motherboard at the same time. Seperate and unique and non-shared IRQ's. Second card had problems dialing out and choppy voice. Swapped the two cards on the motherboard between the exact same slots, both now both work just fine. Lyle - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 01, 2005 4:34 PM Subject: RE: [Asterisk-Users] Qs about FXO/FXS cards What exactly are people seeing when they have issues with their TDM card? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Saturday, January 01, 2005 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to get the TDM400P with the necessary FXO/FXS boards - can I expect the installation to be somewhat straightforward? Any tips to avoid grief? I think the best that anyone can estimate at this time is that a problem exits of some sort and it might be related to a combination of factors, one of which seems to involve specific motherboard designs. There are far too many people complaining about the same issues with the TDM card, several of which opened cases with digium support and got no response. It should be fairly obvious from the many postings on this list since that card was announced that something is not right, and at least some of those systems have digiums T1 cards on the exact same pci bus that are operating just fine. If you read through the archives, you'll see a number of people flapping their jaws using adjectives and adverbs about what they think the issue happens to be, but the majority (if not all) don't have a TDM card and apparently wouldn't touch one with a ten foot pole. What is obvious at this point is: a. no one on the -user list is going to fix (or even hint with any degree of authority) the root-cause b. don't ever post anything to the -dev list regarding a TDM card as that is NOT the forum for digium cards or drivers, c. digium support is not addressing the issue, and, d. the amount of effort required to support the TDM card (stop *, restart the drivers, start *) in its present condition is far greater then what any reasonable non-technical customer will endure. Since this has been an on-going battle, I'd suggest avoiding the TDM card totally, or, take the 50-50 risk to see how high you can raise your frustration level. Would it make more sense to get 3 cheap X100P's and use some kind of ATA for the FXS? Will obviously save a whole bunch of money, but will there be significant added complexity? Three x100p's are likely to cause even more issues due to the high level of system interrupts. (Note: digium has removed them from their web site.) Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
[EMAIL PROTECTED] wrote: What exactly are people seeing when they have issues with their TDM card? I have four of them in service, in everyday use--one RD, one home, and two small office. None has given us the least problem, ever. One caveat that might be germane, given the complaints of others on the list: mine do station (FXS) functions only. We use all of them on SuperMicro mobos, which allow use of the APIC-style IRQs. My customers are very happy with them. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
[EMAIL PROTECTED] wrote: What exactly are people seeing when they have issues with their TDM card? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) Since you asked, and since I'm well into this bottle of Merlot on New Year's day: 1. Power alarms. WTF does that mean? Wish I had some support docs. 2. On bootup, Excessive leakage module x, ProSLIC failed Auto Configuration. Again, WTF? Reboot and it's ok. But, just a reboot after driving 100+ miles to the client site is not a good option. 3. On bootup, a LED won't light. When zapata gets to it, it can't find the channel. Usually means a complete power cycle to get it to work. 4. A TDM card that isn't recognized at all. DOA. 5. Impedience matching to eliminate humm? I'm calling Matt on Monday, and hopefully he'll RMA these cards. I hope that everyone that has a life is out enjoying the New Year. Cheers, Mike -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
On Sat, 2005-01-01 at 16:14 -0700, Michael Welter wrote: Since you asked, and since I'm well into this bottle of Merlot on New Year's day: 1. Power alarms. WTF does that mean? Wish I had some support docs. 2. On bootup, Excessive leakage module x, ProSLIC failed Auto Configuration. Again, WTF? Reboot and it's ok. But, just a reboot after driving 100+ miles to the client site is not a good option. 3. On bootup, a LED won't light. When zapata gets to it, it can't find the channel. Usually means a complete power cycle to get it to work. Those first 3 all sound like you have a problem with power supply and consistency. You don't mention what modules you have in the cards, but I bet you have FXS ports and have too light of a power supply for the job. 4. A TDM card that isn't recognized at all. DOA. 5. Impedience matching to eliminate humm? I'm calling Matt on Monday, and hopefully he'll RMA these cards. I hope that everyone that has a life is out enjoying the New Year. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
One rather common problem (which started the most recent thread on the subject) is the card simply fails to process pstn-fxo calls. Most seem to suggest it happens about once per week or two. When it fails, reloading the drivers clears the problem (which requires taking * down to do it). There are no log messages to hint at why. Another problem is documented in bug #2023 (and 2022), which describes voicemails left via a pstn-tdm call are consistently very low volume. What exactly are people seeing when they have issues with their TDM card? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Saturday, January 01, 2005 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to get the TDM400P with the necessary FXO/FXS boards - can I expect the installation to be somewhat straightforward? Any tips to avoid grief? I think the best that anyone can estimate at this time is that a problem exits of some sort and it might be related to a combination of factors, one of which seems to involve specific motherboard designs. There are far too many people complaining about the same issues with the TDM card, several of which opened cases with digium support and got no response. It should be fairly obvious from the many postings on this list since that card was announced that something is not right, and at least some of those systems have digiums T1 cards on the exact same pci bus that are operating just fine. If you read through the archives, you'll see a number of people flapping their jaws using adjectives and adverbs about what they think the issue happens to be, but the majority (if not all) don't have a TDM card and apparently wouldn't touch one with a ten foot pole. What is obvious at this point is: a. no one on the -user list is going to fix (or even hint with any degree of authority) the root-cause b. don't ever post anything to the -dev list regarding a TDM card as that is NOT the forum for digium cards or drivers, c. digium support is not addressing the issue, and, d. the amount of effort required to support the TDM card (stop *, restart the drivers, start *) in its present condition is far greater then what any reasonable non-technical customer will endure. Since this has been an on-going battle, I'd suggest avoiding the TDM card totally, or, take the 50-50 risk to see how high you can raise your frustration level. Would it make more sense to get 3 cheap X100P's and use some kind of ATA for the FXS? Will obviously save a whole bunch of money, but will there be significant added complexity? Three x100p's are likely to cause even more issues due to the high level of system interrupts. (Note: digium has removed them from their web site.) Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems to use asterisk with mysql /odbc
hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find debugging-options to output invalid login-passwords. Ok, i have made the following: debian is my OS. mysql is installed and working. i has compiled astersk as follows: Modefying: /usr/src/asterisk-1.0.3/channels/Makefile USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 make, and: make install are correctly. i have probed many choises: chois1: i has create a database sipfriends: mysqladmin create sipfriends the database: CREATE TABLE Sipfriends ( Name varchar(40) NULL default '', Secret varchar(40) NULL default '', Context varchar(40) NULL default '', Username varchar(40) default '', paddr varchar(20) NULL default '', Port int(6) NULL default '0', Regseconds int(11) NULL default '0', PRIMARY KEY (Name) ) TYPE=MyISAM; i setting up: /usr/asterisk/etc/asterisk/res_odbc.conf [mysql] dsn = sipfriends username = root password = pre-connect = yes it is not sql-password set. i have only access to this machine. asterisk can't authenficate users from the database. chois2: i copy from asterisk-sources: contrib/scripts/retrieve_sip_conf_from_mysql.pl in my /usr/asterisk/etc/asterisk directory. i create a mysql database: mysqladmin create sip i pasted the table: CREATE TABLE sip ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); in the database. i add the following line in sip.conf: #include = retrieve_sip_conf_from_mysql.pl asterisk say: == Parsing '/usr/asterisk/etc/asterisk/sip.conf': Found == Parsing '/usr/asterisk/etc/asterisk/= retrieve_sip_conf_from_mysql.pl': Not found (No such file or directory) mysql auth is not working in asterisk. what can i do please? thankx for your help --- tel : 089 2500 7676 homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
[EMAIL PROTECTED] wrote: On Sat, 2005-01-01 at 16:14 -0700, Michael Welter wrote: Since you asked, and since I'm well into this bottle of Merlot on New Year's day: 1. Power alarms. WTF does that mean? Wish I had some support docs. 2. On bootup, Excessive leakage module x, ProSLIC failed Auto Configuration. Again, WTF? Reboot and it's ok. But, just a reboot after driving 100+ miles to the client site is not a good option. 3. On bootup, a LED won't light. When zapata gets to it, it can't find the channel. Usually means a complete power cycle to get it to work. Those first 3 all sound like you have a problem with power supply and consistency. You don't mention what modules you have in the cards, but I bet you have FXS ports and have too light of a power supply for the job. Oddly, the maximum power requirements of the TDM400 (fully loaded with 4 FXS modules) is 20W. That'd have to be a pretty weak power supply (or heavily loaded chassis) to have problems drawing that power. Still, I agree that the power supply is a suspect. I'd want to know who makes the power supply, which model it is, and whether that model has a good reputation. An electrically noisy power supply could cause the kinds of anomalies described. So could a faulty supply, of course. More important to my mind is the overall quality of the power feeding the system. Is a dedicated electrical circuit employed? Isolated, insulated grounding conductor right back to a separately-derived source? Power conditioner? So many of the problems people are having with the TDM cards sound like power-quality issues, one has to wonder. I don't mean that as a panacea, because the TDM400 troubles seem to go beyond any one issue. It's merely one thing that might bear looking into. It'd be nice to see some statistics on not only what percentage of TDM400 users are having problems, but also what kind of environment they're in. I'd want to know about the elctrical environment, manufacturer and model of each system component (power supply and motherboard especially). I'd also like to get a report from a circuit analysis performed on the PSTN loop. I realize that much of this would be impossible to get, but one of the most important steps towards solving a bug is being able to identify the conditions which cause it. So far that data is not known, which is a large part of the reason the problem is not getting fixed - no one knows exactly what is causing the troubles - we just have symptoms. What if, for example, the TDM400 issues were a cumulative thing? If you had over 6dB of attenuation on the PSTN loop, coupled with greater than 5V potential on the neutral-ground of your elecrical receptacle, compounded by a cheap power supply, exascerbated by a Via-chipset, would you not be virtually guaranteed some strange behaviour? But if your PSTN was -3dB, your electrical feed derived from a power conditioner, your power supply manufactured by PC Power Cooling, and a ServerWorks chipset-based MoBo, would your system always be faultless? With enough data, we could really start to hone in on this animal. 4. A TDM card that isn't recognized at all. DOA. 5. Impedience matching to eliminate humm? I'm calling Matt on Monday, and hopefully he'll RMA these cards. I hope that everyone that has a life is out enjoying the New Year. http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
Rich Adamson wrote: Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. Except for the little problem I've fought for about a week without any Joy - no combination of efforts from numerous sources (wiki, this forum members, my efforts) has succeeded in a spa-3000/asterisk combination that actually works. If you have specific spa-3000 and asterisk configs that actually work with both spa-3000 ports I'd sure like to have you share them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
Steven Critchfield wrote: Those first 3 all sound like you have a problem with power supply and consistency. You don't mention what modules you have in the cards, but I bet you have FXS ports and have too light of a power supply for the job. I'm not at the client sites, but my test system BIOS reports (with a TDM22B installed): VCORE 1.676V DDR Vtt 1.344 +3.3V 3.28V +5V 4.945 +12V12.544 5VSB4.945 The other card is also a TDM22B, and he DOA card is a TDM40B. I've rotated all cards throught my test system with varying degrees of flakines. Cheers, -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
Except for the little problem I've fought for about a week without any Joy - no combination of efforts from numerous sources (wiki, this forum members, my efforts) has succeeded in a spa-3000/asterisk combination that actually works. If you have specific spa-3000 and asterisk configs that actually work with both spa-3000 ports I'd sure like to have you share them. I have managed to get it work, though I don't use it now. You set up some random SIP account on your * server and feed that authentication information into the PSTN Line VoIP settings. You then enable the PSTN-to-VoIP gateway, set PSTN Caller ID Pattern to *, then set call-forwarding under PSTN User to: Cfwd Sel1 Caller: * Cfwd Sel1 Dest: 123 where 123 is an extension in the context that the SIP account on the * server is in. -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@home ISO install of ISDN card with HFC ?
Hi Have anybody successfully installed ISDN with HFC chips on [EMAIL PROTECTED] ISO ? Please tell me how you did it ? Thank you ! HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip reload - Hang
Sorry about not including the additional info (I realized right after I sent the message of my mistake): Fedora Core 3 (from ISO images); no updates applied Asterisk from CVS (as of this morning) Zaptel from CVS Libpri from CVS No extra Patches No Extra modules Hardware is a Shuttle SS51G w/ Intel Celeron 2.4 GHz; 512 MB RAM; 80 GB Western Digital hard drive. DHCP Assigned address from my main server (address is actually static based on MAC address) No firewall is active I'm more than happy to run gdb to see where the lockup is, but I'm a bit rusty on how to debug a multi-threaded app using gdb. Thanks in advance for any assistance. -- Scott Gruby mailto:[EMAIL PROTECTED] http://www.gruby.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio breakup problems
I've been having audio breakup problems (on my end) in my Asterisk tests. I'm not sure of the most likely source of this quality problem. 99% of my LD calls are calling into a tele-conference service called freeconference.com for group meetings. Its a free phone conference system that works quite well with pstn phones. I've been using it for quite some time. But the audio problems after setting up Asterisk and VoipJet are now unacceptable. About 30 minutes into the call I have difficulty understanding what others are saying because their voices are breaking up. Short calls seem to work fine, quality is good. I'm using Asterisk 1.0.3 on Whitebox Linux (still a novice) I'm using: - Analog phone with Sipura SPA-1001 adapter - VoipJet for termination. - ulaw codec - Comcast cable modem connection Can someone help me narrow down where to begin to solve this? Should I try another termination provider? SPA-1001 settings? Different Asterisk settings? Should I provide my Asterisk conf files here? Is a teleconference service like freeconference.com more likely to have this type of issue than standard phone calls? Thanks. -- Lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
Nabeel Jafferali wrote: Except for the little problem I've fought for about a week without any Joy - no combination of efforts from numerous sources (wiki, this forum members, my efforts) has succeeded in a spa-3000/asterisk combination that actually works. If you have specific spa-3000 and asterisk configs that actually work with both spa-3000 ports I'd sure like to have you share them. I have managed to get it work, though I don't use it now. You set up some random SIP account on your * server and feed that authentication information into the PSTN Line VoIP settings. You then enable the PSTN-to-VoIP gateway, set PSTN Caller ID Pattern to *, then set call-forwarding under PSTN User to: Cfwd Sel1 Caller: * Cfwd Sel1 Dest: 123 where 123 is an extension in the context that the SIP account on the * server is in. I've been down that road - Asterisk reports an error (see the forum history for the exact error message as my server is currently offline awaiting a replacement TDM400 card). If you and other folks who have this working would manage to screen shot the exact sipura configuration - all pages (just so no little forgotten tweak gets by) and the sip.conf and extensions.conf sections I'll give this another go. Hmmm, silly me, error message was in outbound email queue - so here it is again: Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1 Have you tried the A prefix trick, which uses Line 1 Call Forwarding as opposed to PSTN Line Call Forwarding (with the added advantage that the SPA-3000 does not pick up the SPA-3000 line until after the extension/* picks up)? -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
I have experienced nothing but grief when trying to set up the PSTN part of the SPA-3000. Everything from crackly audio to fast busies. BTW I take that back. I spent an hour on this after posting my last email, and with a little tweaking, everything seems to be working well now. -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
Nabeel Jafferali wrote: Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1 Have you tried the A prefix trick, which uses Line 1 Call Forwarding as opposed to PSTN Line Call Forwarding (with the added advantage that the SPA-3000 does not pick up the SPA-3000 line until after the extension/* picks up)? Yep - in fact the above error message used to have an A in front of the 714 - found out that basically anything in that field would cause the immediate forward... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
Rich, I have been wondering if the spa 3000 would make a good PSTN interface for an * box where POTS is the only available (or practical) service. Have you implemented this? Are there any limitations or known issues? The SPA2000 sure seems to work well as an ATA, even had good luck with fax over IP using g.711 and the fax detection in zaptel and the SPA (turns off echo cancel dynamically when the CNG tone is heard I believe). Can you use the FXS and FXO ports at the same time, for two separate calls via * ? The SPA 3000 is small enough that a half dozen of them would be manageable, any more than that and your are usually in the T1 price range for service anyways. Thanks, Damon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, January 01, 2005 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards snip Current experience... three spa-3000's are far more stable then a TDM card, and you'll get three fxo's plus three fxs's for less money. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
I hate to ask the obvious But what's your power quality like? Is the system on a UPS? UPS supplied power makes a huge difference in system stability. I wouldn't run a server for anything (including testing) without it. Second, what class of hardware? You do get what you pay for and flakiness can often be traced to power issues. From what I can tell the Digium hardware does some signal processing magic by relying heavily on system power and cpu power. The first clue here is the 12v plug to provide dial tone/ring to your ATAs. BTW, Ring on a analog phone is typically 90vAC. Dial tone is @48V. So if you put a bunch of analog devices in you are begging for headaches. I learned those numbers after being shocked. I don't strip phone wire with my teeth anymore. Shame on me for being lazy. I'm a firm believer in not running production systems on bargain hardware. I had nothing but grief out of my desktop class and generic trash systems. And yes, Shuttle is generic trash, as is ASUS, and A Open and a host of other Taiwan Special stuff. The way you save money in those systems is by making thinner PCB's which will drive you insane trying to troubleshoot. One tweak of your case and you can lose some contacts. At any rate, judge a circuit by it's thickness. Trash is like paper and flexes. Quality is thick and will cut you before it bends. I'm running older, but solid hardware and not seeing any issues. I'm using a Compaq Proliant 1850R Gen1 dual PII 400 with 512MB ram, GB ethernet, and SATA Hardware RAID. Cheap, efficient, redundant. And for a Debian box, good enough. Initial testing with TOP shows that one ATA to VOIP connection costs 4% of CPU to start up and then 2% to carry. Considering we have 10 handsets and 10 employees with 4 lines and normally no more then 2 people on the POTS lines I think we're in good shape. If you're planning to run a E*trade call center, you may want more substantial hardware. If you are planning the MomPop Voicematrix @Home you may be just fine with a old Proliant. They have redundant power supplies and they are cheap and indestructible. Although it's a bit loud to keep in the bedroom. :) Don't get me wrong, I'm not trashing your hardware. If you can run cheap bargain hardware and get it to work great. But my experience has been that I lost my A** on generic knockoff stuff when I sold PC's for a living. I spent a lot of time chasing errors that I never could find the cause of. Granted my webservers run Windows... And this is a linux app... But I see uptime in the range of months with Proliant hardware. That *is* remarkable for MSFT products. Anyhow that's my two cents. I wonder if there is a correlation between hardware class and issues with TDM boards? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Saturday, January 01, 2005 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards [EMAIL PROTECTED] wrote: What exactly are people seeing when they have issues with their TDM card? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) Since you asked, and since I'm well into this bottle of Merlot on New Year's day: 1. Power alarms. WTF does that mean? Wish I had some support docs. 2. On bootup, Excessive leakage module x, ProSLIC failed Auto Configuration. Again, WTF? Reboot and it's ok. But, just a reboot after driving 100+ miles to the client site is not a good option. 3. On bootup, a LED won't light. When zapata gets to it, it can't find the channel. Usually means a complete power cycle to get it to work. 4. A TDM card that isn't recognized at all. DOA. 5. Impedience matching to eliminate humm? I'm calling Matt on Monday, and hopefully he'll RMA these cards. I hope that everyone that has a life is out enjoying the New Year. Cheers, Mike -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
On Sat, 1 Jan 2005 15:52:50 -0500, Nabeel Jafferali wrote: Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to get the TDM400P with the necessary FXO/FXS boards - can I expect the installation to be somewhat straightforward? Any tips to avoid grief? Would it make more sense to get 3 cheap X100P's and use some kind of ATA for the FXS? Will obviously save a whole bunch of money, but will there be significant added complexity? Here's an option that some might consider, especially in light of the ongoing problems with virtually all small FXO interfacesthe Zultys 4x5 SIP phone. This SIP phone includes an onboard 4 port router wit QoS and an FXO interface. At point of introduction early in 2004 the FXO i/f was only used as a lifeline. The firmware setup allowed the suer to pass local calls to the FXO while passing all other calls to * via SIP. Alternatively, the phone could try SIP calling outboard before falling back to the FXO. However, about two weeks ago Zultys released firmware that makes the FXO available as a SIP reource to *. Calling coming in on the FXO can be routed to * for transfer or VM purposes. It's a little strange since we're accustomed to having the FXO/FXO i/fs on the server. However, you could bring the POTS lines to the desktops and into the 4x5 phones. Then have the call pass to * when required. BTW, the router functions such as DHCP etc can be defeated if desired. Also, the 4x5, unlike some phones, requires only one registration to support multiple call appearances. From on registration the 4x5 support 4 active SIP calls and one on the FXO, at the same time. Michael P.S. - I'm buying one of these from a friend who doesn't need his anymore. I had it on loan for a month back in the summer. It worked well with* but the FXO * firmware was not yet available. -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing call (Sip phones to PSTN)
Hi All,
Everytime I make outgoing call, the channel at TDM
card doing hungup after might be a second the
destination number get ringing.The call is from sip
phones to PSTN phone. The sip phones was completely
registered to asterisk. here is my conf :
sip.conf :
[1234]
type=friend
username=1234
secret=
host=dynamic
context=sip-ph
extensions.conf :
[sip-ph]
exten = _NXX,1,Dial(Zap/g1/${EXTEN}|10,t)
zapata.conf:
signalling=fxs_ks
group=1
context=incoming
channel=3-4
I thought in this case the context sip-ph doesn't
need to exist in zapata.conf,does it ?
So where is my mistakes ? Please help
thanks,
ron
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[Asterisk-Users] Announcements via IAX phones
Hello-- What I'd like to do: Use IAX softphones running on computers, in Auto-answer mode, with sound going to speakers, as a sort of public announcement system. What isn't working: Well, my first experiment was to set up the MeetMe system described on the Wiki... This works fine for voice announcements. You pick up a phone, dial the right extension, and an agi is fired up to put files in the call spool to call the autoanswer extensions, simultaneously as it were, and all are entered into the same conference. The caller is the admin. You speak, they hear. It works fine. I changed the kicked gsm to a beep, as the conference is terminated by kicking everyone off, and it is kinda comical to end an announcement with You are kicked from the conference message at the end... But, I need to play automated announcements. So, I whipped together an agi to generate the sounds in the right sequence. But, how do I link it to a conference? Since they are not ZAP channels, the softphones don't seem to be able to handle the background agi option (the Meetme 'b' option), which would have been a potential way to play the sounds to the conference. I tried a Meetme call inside the agi. It kinda hangs-- You can be in a conf, but you can't play sounds to it until it ends... that won't work! Thought about firing an AGI to call each softphone and dump the sequenced list of sounds, but it would take a while to run serially thru the list of phones to announce. This is unacceptable. It's all at once, or it's useless. Anyone have any ideas? I'm running short on them at the moment. Seems asterisk is powerful enough to do anything... but this? murf -- Steve Murphy [EMAIL PROTECTED] Electronic Tools Company ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
On Sat, 2005-01-01 at 17:25 -0700, Michael Welter wrote: Steven Critchfield wrote: Those first 3 all sound like you have a problem with power supply and consistency. You don't mention what modules you have in the cards, but I bet you have FXS ports and have too light of a power supply for the job. I'm not at the client sites, but my test system BIOS reports (with a TDM22B installed): VCORE 1.676V DDR Vtt 1.344 +3.3V 3.28V +5V 4.945 +12V 12.544 5VSB 4.945 And from BIOS you for sure are not loading a driver and you aren't having to drive the ringing voltage. Part of my concern on power supplies is that I have abused them for non standard functions and know that many of them will pulse the 12v lines and probably the 5v lines as well if you draw too much power. So consider the option that your server is running smooth with no activity and the hard drive(s) spool down. Then a call comes in and asterisk goes to trying to write to the disk for logging as well as generate ring voltage. This could cause a low quality PSU to see a spike that it isn't capable of handling and it would pulse and there is your power alarm. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
