RE: [Asterisk-Users] odd behavior - adtran ta 850 + t100p
I've never used an 850, but I had similar problem on the 750 when I had the channel configured wrong in the 750 console. Have you tried reseting the config and making sure everything is FXS Loopstart. Also, have you tried another AMP-50 cable with your bank. I had a bad cable that was crossing signal with all channels. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeff Roberts Sent: Thursday, July 08, 2004 11:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] odd behavior - adtran ta 850 + t100p [EMAIL PROTECTED] wrote: I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked to module 2 of the same fxs card then asterisk (correctly) only sees that module go off hook. When plugging a phone into any of the fxs pairs, I only get dial tone for a second or two and then I get silence. However, I can still dial extensions and get through. I'm not sure but maybe it is a config problem with the ta 850, as it takes a little more manual configuration than the ta 750 I worked with before. Anybody have any pointers? Here is the output on the console when I pick up a phone on module 1, and module 2, respectively: [EMAIL PROTECTED]:~# asterisk -r Asterisk CVS-HEAD-07/06/04-12:37:58, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-HEAD-07/06/04-12:37:58 currently running on slack1 (pid = 702) - Remote UNIX connection -- Starting simple switch on 'Zap/5-1' -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' -- Starting simple switch on 'Zap/8-1' -- Hungup 'Zap/5-1' -- Hungup 'Zap/6-1' -- Hungup 'Zap/7-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/5-1' Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/8-1' -- Hungup 'Zap/5-1' -- Hungup 'Zap/6-1' -- Hungup 'Zap/7-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/6-1' -- Hungup 'Zap/6-1' Here is zaptel.conf: span=1,0,0,esf,b8zs loadzone = us defaultzone=us fxsks=1 fxoks=5-24 And here is zapata.conf: [channels] transfer=yes context=default language = en usecallerid = no hidecallerid = no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no signalling=fxs_ks echotraining=yes group = 0 channel = 1 context=trusted group = 1 signalling = fxo_ks rxwink = 300 channel = 5-24 Any help appreciated, -Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well I tried setting up the the unused fxo ports, tried setting them to unused, and even moved the fxs cards around in the bank to see if it would make any difference. No joy though. Anybody know how to run some self tests on this bank to be sure its the problem? I'm pretty sure adtran will fix or replace the bank, but I'm sure they are going to want me to explain the problem but I'm not sure what info they'll need. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New PBX Help
First , you need to see what your insurance policy covered. If it covered replacement, then the easist thing for you to do is make the claim and replace your old pbx through a local service provider(asterisk or not). Second if you know next to nothing about pbx's and phone, then the time it takes you to learn asterisk, or whatever you choose, means no phones for your company's employees which in turn could equal more lost revenue, etc. Depending on your familiarity with linux, the learning curve could be steep and prove frustrating considering everything else you'll be dealing with (new network infrastructure, new computers, new servers, new telco/data circuits). Less expensive components does not always equal cheaper. Before I installed my system I knew tip/ring and some T-1 stuff on the telco side. It took me 3-4 months to get completely comfortable with asterisk and all the other telco things before I deployed my asterisk system, which replaced a working legacy pbx. The most difficult thing was the telco side. There are many ways to get dialtone, and telco engineers aren't always forthcoming with information. They are used to dealing with vendors that know what they know. my $0.02 -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Wagner Sent: Wednesday, July 07, 2004 9:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New PBX Help Hi All, We recently had an old office building burn down. The office housed maybe 20-30 people. Only about 10 or so of those had their own extensions. We had a standard pbx from an area communications company, and I'm not quite sure about what kind of phone lines were there, I only know that their were actually 3 phone numbers, but everyone could get an outside line if they needed to. We're looking at moving to a new building, and I would like to use Asterisk, because I feel it would be cheaper than purchasing a pbx. Is there any reccomendations as to how I might set this up??? Keep in mind that I know next to nothing about pbx's and phone systems. Any help is greatly appreciated. Thanks! -Mike Wagner MCCESC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help, Ideas and Ready for use Solutions
If there is already an existing phone system in place, you could easily migrate to an asterisk based solution if your internal phones are analog. The big question for you is not number of phone lines, but peak utilization. Here's what I have. 141 Analog Phone Lines 15 SIP IP Phones (Mix Cisco 7960 and Polycom IP500) T-1 EM For Long Distance PRI For Local Calls with 200 DID Numbers. Max concurrent calls 15-20 (30-40 active channels) I serve all of this from a machine with the following config Dual Xeon 2.4 Ghz 2 Gig Memory 3x36 Gig SCSI Drives in Hardware Raid 5 Configuration. 2xT400P 6xAdtran TA750 Channel Banks with 6xQuadFXS cards in Loopstart Config. This is more than enough for my call volume. The CPU utilization is minimal. As you need more functionality at each phone location, you could switch to IP phones as needed. If you are starting from scratch, you will have a higher startup cost in equipment. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: Friday, June 04, 2004 9:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help, Ideas and Ready for use Solutions Hi, I would like to ask you for advice how to solve the following case: I have a client (who happened to be my friend) and I have convinced him that the IP PBX solution is much better than the conventional telephone centrals (PBX). At the beginning he wanted to buy PBX Panasonic, but at this moment he is waiting for my decision. Because at the moment we are not so deeply familiar with these technologies to be able to offer him all at once, we need your help. My client wants to have 100 internal telephones, and between 10 (existing analog lines to PSTN) and 30 (E1/PRI ISDN) external telephones. Is there any ready solution for this case we could use and how much it will cost? Thank you in advance. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
I've got Asterisk STABLE-CVS-4/19/04 with 12 Cisco 7960 phones 6.0 Firmware using ulaw, 6 Polycom IP500 ulaw phones, and 192 Zap channels. I have Gig-E Copper to my server and 100Mbit-Full to all my phones. I haven't had any choppy audio at all. My switch is a Cisco 4500. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tom Sent: Friday, May 07, 2004 11:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) At 09:43 AM 5/7/2004, you wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro server. Analog phones through our TDM400P do sound much better but the audio problems on our Cisco SIP phones are echo problems. People are working on solutions. Tom Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Goryachev Sent: Wednesday, April 21, 2004 2:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE) Should this actually attempt more than a single ping before claiming the remote is unreachable? ie, one packet (out of the two - one request + one reply) might be lost or intermittent congestion might be involved. Perhaps a config option for setting number of consecutive ping requests are un-responsive. Also, subsequent requests might be sooner than otherwise queued. ie, successfully answered probes are re-sent every 60 seconds, while after an un-successful probe, we re-send the next probe in 10 seconds Just my 0.02c worth On a somewhat related note. I was experiencing some random SIP UN/REACHABLE messages during random points during the day. This would also come hand-in-hand with poor SIP call quality (jitters, stutters, etc). Yesterday I was tryint to debug a choppy SIP phone and it just so happened that I was in my lab , and noticed that we were using Ghostcast server over multicast to send images to some new PCs. On a whim, I cancelled the ghostcast session and the problem immediatly vanished. Must be a misconfig on the switch (Cisco Cat 4500 with all copper 10/100/1000 ports ) cause the switch load was minimal, but somehow the multicast traffic was screwing with the SIP transmission over the wire. Just something for other people to look for. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P / ZAP / PRI errors
In laymans terms. To use your telco's T-1 as the timing source span=1,1,0,esf,b8zs,yelllow To use the internal clock of the card you would use (I'm pretty sure that this would only be used for channel banks, or connections to other PBX hardware. I don't think a telco is going to use your PBX as a timing source) span=1,0.0,esf,b8zs,yellow If you have multiple telco connections on multiple spans you would have something like this span=1,2,0,esf,b8zs,yellow(secondary time source) span=2,1,0,esf,b8zs,yellow (primary time source) span=3,0,0,esf,b8zs,yellow (provide the time source, i.e. channel bank) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, April 12, 2004 9:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T100P / ZAP / PRI errors Now you've got me utterly confused ... So, in layman's terms, if I connect a T100P to a circuit provided by the Telco, and the Telco says that they will provide timing, I have to put WHAT? span=1,0,0,esf,b8zs,yellow this means '0' this span is not a sync source, i.e. the Telco will provide my 8kHz. Could one use '2' with impunity (span=1,2,0,...)? I am still not clear under which circumstances one should use '0' versus '2'. WW - Original Message Follows - Holy crap people, trim your replies! You didn't say what's at the other end of your PRI line, but you might try having the other end be the timing sync source. Try: span=1,0,0,esf,b8zs instead. Maybe that will help. We need to get this documented *clearly* once and for all. Zaptel T1/E1 hardware either free-runs to its own internal 8kHz time source, or it tries to lock to the recovered clock from the line. Zapata.conf says that timing of 0 means do not use this span for timing. Zero does not mean slave timing, it means not to use this span as a recovered clock source for timing at all. Timing values of 1 or 2 mean try to lock the internal clock to the recovered clock from the span. A value of 0 means that this span's recovered clock never gets used as a timing source. A value of 1 means that this span is the primary clock source -- If the span is up, try to lock the internal clock to the clock recovered from this span. A value of 2 means to use this span for timing only if the primary span is down. To reiterate: a value of 0 means that the other end must be locking to the zaptel's clock or else clock slips will occur. Feel free to correct me if I'm wrong, but I am pretty sure I have this right. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P / ZAP / PRI errors
I'm running Zaptel CVS from April 8, LibPRI CVS April 8, and v-1.0 CVS April 7. With dual T400P cards with no PRI errors at all. Possibly something driver/config related? Are you timing from your PRI? I remember getting some PRI errors when my timing config was hosed. Could you post your zaptel.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Monday, April 12, 2004 12:09 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors Mike- Do you have access to any kind of PRI test set, like a T-Bird or something. A red alarm would be easy to capture I imagine - it would be nice to confirm that it's a problem particular to your site. The reason I mention software is that I've noticed a lot of other messages regarding these spurious alarms that immediately clear. I've noticed changes in the code having to do with the timing source. But maybe you're right about the telco doing something odd! Cheers Scott www.evtmedia.com -Original Message- From: Mike Sturdee [mailto:[EMAIL PROTECTED] Sent: Monday, April 12, 2004 9:00 AM To: Scott Stingel Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors I have been seeing this for over a month, and blaming it on our generally incompetent telco, so it's definately not a new issue. On Mon, 12 Apr 2004, Scott Stingel wrote: This doesn't appear to be a load issue, since normally in that case I would expect you would get a lot of (usually harmless) frame reject messages in your /var/log/asterisk/messages file, and perhaps an occasional missed/double interrupt message on the console. I wonder if there have been new bugs introduced in the PRI code. I've seen a lot of changes in the timing section of the code at least on the dev list. By all means, report it directly to them (a phone call is best). Cheers Scott Stingel www.evt.media.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sturdee Sent: Monday, April 12, 2004 8:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] T100P / ZAP / PRI errors My PRI is being reset at least once a day with the following errors in the logs. zaptel, zapata, libpri, and asterisk are from CVS this morning.. This has been happening for weeks on all versions (including -stable). the T100P card appears to NOT be sharing an IRQ. xenon# cat /proc/interupts CPU0 0:1203977 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 12004595 XT-PIC t1xxp 8: 1 XT-PIC rtc 9:1046347 XT-PIC eth0 14: 21317 XT-PIC ide0 NMI: 0 ERR: 0 / Is this something I should be seeking support from Digium on being their card? Apr 12 11:04:59 WARNING[1226062640]: PRI: Short write: -1/15 (Unknown error 500) Apr 12 11:04:59 WARNING[1226062640]: Detected alarm on channel 1: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 2: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 2 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 3: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 3 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 4: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 4 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 5: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 5 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 6: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 6 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 7: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 7 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 8: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 8 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 9: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 9 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 10: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 10 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 11: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 11 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 12: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on channel 12 Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on channel 13: Red Alarm Apr 12 11:04:59
RE: [Asterisk-Users] dreaded Caller*ID failed checksum
Did you install the micro filters that came with with your ADSL modem. Usually you get 3-4 of these. They are used to protect your analog lines from the additional signal noise from the ADSL signal. -sb Radio Shack item number 279-103 for about $15 each -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Oliver WilcockSent: Thursday, April 08, 2004 10:30 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] dreaded Caller*ID failed checksumSome input for those with bigger brains like, perhaps, Jeremy Hall, to ponder in relation to the Caller*ID failed checksum message. 1. I have a Digium X101P (and TDM420P) 2. I installed in January and had no problems with Caller ID 3. I played with Asterisk including a few CSV updates in February (stable 1.0, I think). 4. Then I had problems with Asterisk never reporting Caller ID to phones on the TMD420P and returning the "failed checksum" message sometimes. 5. I updated to Apri 4 CSV of zaptel and asterisk and now Caller ID works. 6. I have ADSL, which introduces noise on the line and I have an emergency phone which bypasses Asterisk (in case of power failure). That phone reports Caller ID without problems. 7. Now the X101P fails to detect hangups reliably, though I think it is more reliable than it was in March when I was using the stable code.
[Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI-T400P-Asterisk-T400P-Adtran 750(L36 Firmware)-RAS Server. I have 4 Zap channels signalled FXO_KS to the 750 with FXS_LS channels, On-Hook messaging disabled, the rest defaults for the channels. In zapata.conf I've tried with both busydetect=yes and busydetect=no busycount=6, busycount=10, callprogress=yes, callprogress=no all combinations. The weird thing is, that if I forward the incoming call from the PRI out another channel on the PRI into a POTS line hooked into the RAS server, the connection is fine. In my view, that rules out the PRI and points the blame at either how the adtran is configured, or the how the channel itself is configured. Can anyone with a _working_ configuration similar to this chime in with some config info on the Zap channel and the channel bank config? Thanks in advance. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, April 07, 2004 10:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750 Bisker, Scott (7805) wrote: I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI-T400P-Asterisk-T400P-Adtran 750(L36 Firmware)-RAS Server. I have 4 Zap channels signalled FXO_KS to the 750 with FXS_LS channels, On-Hook messaging disabled, the rest defaults for the channels. In zapata.conf I've tried with both busydetect=yes and busydetect=no busycount=6, busycount=10, callprogress=yes, callprogress=no all combinations. We have 4 750's and one TSU 600 working with PC anywhere for data communications for our support department. We have on this system 2 T400P's. The only thing I can say is who are you getting your timing from. We are able to get modem calls and faxes without problems. But this is only using PRI from Allegenice. We also have a LD service T1 from Sprint that is in no way able to handle any data calls. Our Adtrans are out of the box without any changes to them. This is our settings in our zapata.conf. ; Enable echo cancellation echocancel=yes ;echocancelwhenbridged=yes immediate=no ;adsi=yes usecallerid=yes hidecallerid=no ;callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes musiconhold=default signalling = fxo_ks Hope this helps. The weird thing is, that if I forward the incoming call from the PRI out another channel on the PRI into a POTS line hooked into the RAS server, the connection is fine. In my view, that rules out the PRI and points the blame at either how the adtran is configured, or the how the channel itself is configured. Can anyone with a _working_ configuration similar to this chime in with some config info on the Zap channel and the channel bank config? Thanks in advance. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
I'm timing off my PRI from Verizon as well. This is mind boggling. All my Fax machines are fine. The modems connect, but drop the calls after about 1-2 minutes regardless of busydetect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, April 07, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750 Bisker, Scott (7805) wrote: Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? I have firmware L36. Ras card is a Digikey 4 port board on one NT server and others are using the normal serial ports on the servers. The desktops are using there modems connected to there PC's via Serial cables. All our modems are USR Sporters 56K we have about 20 of them. Except for 3 USR Courier 56K. For our fax board we are using BrookTrout I4P on a Windows 2000 server with ZataFax. Everything is working off the timing from the PRI line. Asterisk is older on this installation. This installation is still using .5 from CVS 12/05/03. I belive if it works leave it along! And it works just fine! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap Channels Hang
This could possibly be related to Bug# 0001320 where Zap channels get stuck in a Rsrvd State. I inadvertently put the bug in Zaptel since I had upgraded to Zaptel 0.9.0 the same time I upgraded to asterisk v1-0_stable. When I rolled back to asterisk 0.7.1 with -DOLD_DSP_ROUTINES the problem went away. I'm going to try v1-0_stable with -DOLD_DSP_ROUTINES this weekend to see if the problem goes away. One bad side affect to 0.7.1 is occasional terrible echo on Zap channels. This behavior was not present in v1-0_stable. My $0.02 -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Thursday, April 01, 2004 8:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Zap Channels Hang Luciano, I was having the same thing happen after updating to that code...but since mine is in production I had to quickly go back to the code from two weeks ago. I know it's not a solution...but if you really need it back up now you might want to do that. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luciano Ramos Sent: Thursday, April 01, 2004 6:24 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Zap Channels Hang I am having the some problem here, I had to put a asterisk restart in cron every night. I am running an E100P also, my * ver is Asterisk CVS-02/25/04-20:35:20 Thanks! Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Zap Channels Hang Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/31-1 (default9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network9682908972 )Ring Dial Zap/g2/68290897 Zap/30-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network9938415442 )Ring Dial Zap/g2/93841544 Zap/29-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network9966446872 )Ring Dial Zap/g2/96644687 Zap/28-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network9938716482 )Ring Dial Zap/g2/93871648 Zap/27-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network9686272242 )Ring Dial Zap/g2/68627224 Zap/26-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network9656277802 )Ring Dial Zap/g2/65627780 Zap/25-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/24-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/23-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network9656990622 )Ring Dial Zap/g2/65699062 Zap/22-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network9656763882 )Ring Dial Zap/g2/65676388 Zap/21-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network9626622722 )Ring Dial Zap/g2/62662272 Zap/20-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 9642901182 )Ring Dial Zap/g2/64290118 Zap/19-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network9656276402 )Ring Dial Zap/g2/65627640 Zap/18-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network9642555752 )Ring Dial Zap/g2/64255575 Zap/17-1 (defaults1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 9656990622 ) Up Dial Zap/g2/65699062 Regards, Antonio Rabena __ NOD32 1.700 (20040331) Information __ This message was checked by NOD32 Antivirus System. http://www.nod32.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] Zap channels stuck in 'Rsrvd' state
I've just started having the same problem here today. I did and upgrade over the weekend to Zaptel-0.9.0 and the release candidate for Asterisk-1.0 CVS 3/28/04. I have 6 Adtran 750 FXS_KS for all channels. 1 T-1PRI and one EM_W T-1. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Creel Sent: Monday, March 29, 2004 2:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zap channels stuck in 'Rsrvd' state I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets stuck off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P not ringing.
Title: Message Please post the portion of your dialplan that you are explaining. More than likely you don't have an "r" in your dial command. That lets the calling party hear a ring. e.g. Dial(SIP/1234|20|Tr) -sb -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Mark Messmore, Technical Support, University Telcom Inc.Sent: Monday, March 22, 2004 9:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] T100P not ringing. I posted this problem another time, but with another problem tied in...so let's try this out. I've got an X100P and a T100P on the same box (the x100p was initially for testing, but since it's working fine we are still using it). However, the X100P is tied into a different switch than the T100P is. I won't take the time to explain why...but that's just how things are. ## (T100P) (VoIP) Switch 1Asterisk Box--.--.--.--.--.--.--.--Remote Office | (X100P) | Switch 2 - ## When making an incoming call that's being routed through Switch 1 to the remote office, the call is being routed fine and getting there fine. The phone rings and the person can talk with good clarity. The one problem is that the person making the call from the outside does not hear a ring on the phone. Therefore I'm afraid that people will hang up thinking that it's not getting through. As an FYI when I call the Remote office from the outside and it's being routed through Switch 2 on the X100P I hear ringing just fine. The only clues I found in the mail archive were from 2002 and the suggestion was to update the zaptel driver. I did that, but there was no change. Any ideas would be greatly appreciated. Thanks. Mark
RE: [Asterisk-Users] T100P not ringing.
Title: Message
Nothing is jumping out.
Why
don't you try simplifying your dialplan a little without all the gotos and
includes, and see if you can get it to ring.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.Sent: Monday,
March 22, 2004 1:15 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] T100P
not ringing.
Thanks for the response. Here are two contexts from my
extensions.conf. The number being dialed is in the "bob"
context.
[bob]exten = s,1,Goto(uti-mainst,2450,1)
include =defaultinclude =outboundinclude
=uti-mainst
[uti-mainst]
exten = 2450,1,Dial(SIP/bob,45,Ttr)
include =outboundinclude =bobinclude
=conference
Obviously I've left out a majority of the extensions, etc. But
this should be enough to show you what I have.
Thanks for your help.
Mark
-Original
Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker,
Scott (7805)Sent: Monday, March 22, 2004 12:13 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
T100P not ringing.
Please post the portion of your dialplan that you are
explaining. More than likely you don't have an "r" in your dial
command. That lets the calling party hear a ring.
e.g.
Dial(SIP/1234|20|Tr)
-sb
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.Sent:
Monday, March 22, 2004 9:36 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] T100P
not ringing.
I posted this
problem another time, but with another problem tied in...so let's try this
out.
I've got an
X100P and a T100P on the same box (the x100p was initially for testing,
but since it's working fine we are still using it). However, the
X100P is tied into a different switch than the T100P is. I won't
take the time to explain why...but that's just how things
are.
##
(T100P)
(VoIP)
Switch
1Asterisk Box--.--.--.--.--.--.--.--Remote
Office
|
(X100P)
|
Switch 2
-
##
When making an
incoming call that's being routed through Switch 1 to the remote office,
the call is being routed fine and getting there fine. The phone
rings and the person can talk with good clarity. The one problem is
that the person making the call from the outside does not hear a ring on
the phone. Therefore I'm afraid that people will hang up thinking
that it's not getting through.
As an FYI when
I call the Remote office from the outside and it's being routed through
Switch 2 on the X100P I hear ringing just fine. The only clues I
found in the mail archive were from 2002 and the suggestion was to update
the zaptel driver. I did that, but there was no change. Any
ideas would be greatly appreciated. Thanks.
Mark
RE: [Asterisk-Users] Fuse for Adtran 750 PSU
I got my fuses from a local supplier. Looks like the OEM is Littelfuse. PN: 0481003.V -sb -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jacques LeisySent: Friday, March 19, 2004 10:23 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Fuse for Adtran 750 PSU Sorry for a very stupid question, but I cannot find a supplier anywhere. Where can I buy the 3 Amps GMT fuses for the Adtran's PSU. Car fuse don't seems to fit. What is GTM the abbreviation of Thanks Jacques
RE: [Asterisk-Users] PRI Errors
Update on this. I had the exact same issue today. At almost exactly the same time as yesterday. Possible telco problem? Timing issue with zaptel? Never had this issue before updating libpri as of 3/8. Here's zaptel.conf span 7 is PRI from Verizon, span 8 is T-1 from Sprint. Dual T400P, SMP # span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs span=3,1,0,esf,b8zs span=4,1,0,esf,b8zs span=5,1,0,esf,b8zs span=6,1,0,esf,b8zs span=7,0,0,esf,b8zs span=8,0,0,esf,b8zs em=1-12 fxoks=13-24 fxoks=25-48 fxoks=49-72 fxoks=73-96 fxoks=97-120 fxoks=121-144 em=145-168 bchan=169-191 dchan=192 -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Tuesday, March 16, 2004 11:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI Errors I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Tuesday, March 16, 2004 10:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI Errors Hi Andrew- The unknown error 500 and the frame rejects are somewhat normal - I get thousands of these in a busy IVR system. The underlying cause for these, I think, is that your processor occasionally does not keep up with the frame transmitter on the PRI board - something that will happen from time to time, and asterisk should recover. (although, as previously discussed, asterisk's minimal error handling makes this worse than it should be) The red alarms indicate a loss of synchronisation, or a very high bit error rate. This is a basic problem and should not occur. Are you sure the switch is not shutting down the T1's as part of some housekeeping? Regards, Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew McRory Sent: Tuesday, March 16, 2004 6:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI Errors I've been running Asterisk CVS-02/29/04-12:09:10 for a couple of weeks with no real incidents... LEC-PRI --- T400P - * - SIP/IAX | |- MicroCom ISPorte (faxserver) | |- Max4004 (dialup) This configuration has added some flexibility we didn't have before and I love it but two weeks of uptime the following errors appeared in the logs. All connections were dropped. Is this an * problem or something on the PRI? It happened at 3AM so it wasn't a big deal this time but I hate to see it happen in the middle of the day. === == Mar 14 03:11:54 WARNING[11276]: PRI: Read on 108 failed: Unknown error 500 Mar 14 03:11:54 NOTICE[11276]: PRI got event: 6 on span 1 Mar 14 03:11:54 WARNING[998419]: PRI: Short write: -1/15 (Unknown error 500) Mar 14 03:11:54 WARNING[998419]: Detected alarm on channel 1: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 2: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 4: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 5: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 6: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 7: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 8: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 9: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 10: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 11: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 12: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 13: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 14: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 15: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 16: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 17: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 18: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 19: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 20: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 21: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 22: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 23: Red Alarm Mar 14 03:11:54 WARNING[836629]: PRI: Short write: -1/15 (Unknown error 500) Mar 14 03:11:54 WARNING[836629]: Detected
RE: [Asterisk-Users] PRI Errors
Oh nooo. Completely missed the boat on this one. I was thinking the exact opposite on this. I thought that if set to 1, then the span would _provide_ timing for the connected circuit. My span 1-6 are channel banks and I thought the 1 was providing the timing for the banks, not the other way around I now see my error. I'm suprised I've not had more issues than this. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcin Kuzmicki Sent: Wednesday, March 17, 2004 3:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI Errors Hi, Maybe I'm wrong but you have different oprators - two different switches and you dont synchronize with them you dont use them as your timing source I'd go like like this span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs span=5,0,0,esf,b8zs span=6,0,0,esf,b8zs span=7,1,0,esf,b8zs span=8,2,0,esf,b8zs rgrds Quoting Bisker, Scott (7805) [EMAIL PROTECTED]: Update on this. I had the exact same issue today. At almost exactly the same time as yesterday. Possible telco problem? Timing issue with zaptel? Never had this issue before updating libpri as of 3/8. Here's zaptel.conf span 7 is PRI from Verizon, span 8 is T-1 from Sprint. Dual T400P, SMP # span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs span=3,1,0,esf,b8zs span=4,1,0,esf,b8zs span=5,1,0,esf,b8zs span=6,1,0,esf,b8zs span=7,0,0,esf,b8zs span=8,0,0,esf,b8zs em=1-12 fxoks=13-24 fxoks=25-48 fxoks=49-72 fxoks=73-96 fxoks=97-120 fxoks=121-144 em=145-168 bchan=169-191 dchan=192 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Errors
I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Tuesday, March 16, 2004 10:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI Errors Hi Andrew- The unknown error 500 and the frame rejects are somewhat normal - I get thousands of these in a busy IVR system. The underlying cause for these, I think, is that your processor occasionally does not keep up with the frame transmitter on the PRI board - something that will happen from time to time, and asterisk should recover. (although, as previously discussed, asterisk's minimal error handling makes this worse than it should be) The red alarms indicate a loss of synchronisation, or a very high bit error rate. This is a basic problem and should not occur. Are you sure the switch is not shutting down the T1's as part of some housekeeping? Regards, Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew McRory Sent: Tuesday, March 16, 2004 6:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI Errors I've been running Asterisk CVS-02/29/04-12:09:10 for a couple of weeks with no real incidents... LEC-PRI --- T400P - * - SIP/IAX | |- MicroCom ISPorte (faxserver) | |- Max4004 (dialup) This configuration has added some flexibility we didn't have before and I love it but two weeks of uptime the following errors appeared in the logs. All connections were dropped. Is this an * problem or something on the PRI? It happened at 3AM so it wasn't a big deal this time but I hate to see it happen in the middle of the day. === == Mar 14 03:11:54 WARNING[11276]: PRI: Read on 108 failed: Unknown error 500 Mar 14 03:11:54 NOTICE[11276]: PRI got event: 6 on span 1 Mar 14 03:11:54 WARNING[998419]: PRI: Short write: -1/15 (Unknown error 500) Mar 14 03:11:54 WARNING[998419]: Detected alarm on channel 1: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 2: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 4: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 5: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 6: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 7: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 8: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 9: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 10: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 11: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 12: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 13: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 14: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 15: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 16: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 17: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 18: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 19: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 20: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 21: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 22: Red Alarm Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 23: Red Alarm Mar 14 03:11:54 WARNING[836629]: PRI: Short write: -1/15 (Unknown error 500) Mar 14 03:11:54 WARNING[836629]: Detected alarm on channel 3: Red Alarm Mar 14 03:11:54 WARNING[11276]: PRI: Read on 108 failed: Unknown error 500 Mar 14 03:11:54 NOTICE[11276]: PRI got event: 4 on span 1 Mar 14 03:11:55 WARNING[12301]: PRI: !! Got reject for frame 105, retransmitting frame 105 now, updating n_r! Mar 14 03:11:55 WARNING[12301]: PRI: !! Got reject for frame 105, retransmitting frame 106 now, updating n_r! Mar 14 03:11:55 WARNING[11276]: PRI: !! Got reject for frame 41, retransmitting frame 41 now, updating n_r! Mar 14 03:11:55 WARNING[11276]: PRI: !! Got reject for frame 41, retransmitting frame 42 now, updating n_r! Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 1 Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 2 Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 3 Mar 14 03:11:59 NOTICE[15376]:
RE: [Asterisk-Users] extensions problem (SIP)
In your SIP.conf set callwaiting = no. This will work for single registrations. If you have multiple call appearance on you phone, then it will just ring to the second line (e.g. Cisco 7960). If you only have a single registration, then you should be fine. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Monday, March 15, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] extensions problem (SIP) Jon Lawrence wrote: Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include = dialjon [inboundB] include = dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten = s,1,answer exten = s,2,Dial(SIP/2000,15) exten = s,3,Playback(noone) exten = s,103,Goto(onphone,s,1) snip Am I right in saying: if no one answers at ext 2000 then s,3 is executed. if ext 2000 is busy then 103 is executed. If so then sometihng is wrong. If I'm already on a call, I want 103 to be executed however, this isn't happening. If a new call comes in (whilst I'm talking on ext 2000) I here it ringing on my handset. It depends on your SIP device. Asterisk places the call to your SIP device regardless, since by SIP protocol design the UA is not a slave, it is free. So the SIP ua must answer busy for Asterisk to understand that you're busy. If not, the call is placed to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can limit the number of concurrent calls. There's some code in Asterisk chan_sip.c to limit the number of calls placed to a SIP phone, but right now it's not working at all. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pri Errors, Hanging up Owner
Title: Pri Errors, Hanging up Owner I had the same problem a few weeks ago. I updated to latest zaptel and libpri, and the problem went away. My date is 3/8/04 -sb -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Matthew BrantonSent: Monday, March 15, 2004 2:48 PMTo: Asterisk-Users (E-mail)Subject: [Asterisk-Users] Pri Errors, Hanging up Owner Hey guys, Every so often my pri channels degenerate into a non stop series of Mar 15 06:51:50 WARNING[131081]: chan_zap.c:6263 pri_dchannel: Ring requested on channel 1 already in use on span 1. Hanging up owner. Errors. Anyone else having this problem? I see an old reference to updating your cvs, I am using a fairly updated version, as of say a week ago. Anyone have any experience with this / knows what the problem is? Matt
[Asterisk-Users] ZapRAS over IAX anyone?
I'm just pinging the list for some quick info that I could turn up in google. Has anyone played with doing ZapRAS over an IAX channel? i.e. call comes in T-1 to server 1. Server 1 sends call to server 2 via IAX. Server 2 picksup call with ZapRAS, runs ppp... etc. I don't see why this would be a major issue, just checking to see if it's been done before. Thanks, -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ZapRAS over IAX anyone?
Make that could not turn up in google. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Monday, March 15, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZapRAS over IAX anyone? I'm just pinging the list for some quick info that I could turn up in google. Has anyone played with doing ZapRAS over an IAX channel? i.e. call comes in T-1 to server 1. Server 1 sends call to server 2 via IAX. Server 2 picksup call with ZapRAS, runs ppp... etc. I don't see why this would be a major issue, just checking to see if it's been done before. Thanks, -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk mangling faxes
Michiel,
Are you using WinFax? or one of the Products Based on Winfax? I've seen this on all
of our WinFax Stations, but none of our standalone Fax machines.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of michiel betel
Sent: Wednesday, March 10, 2004 9:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk mangling faxes
Hi list,
Faxes come in over an E1 line (on an TE410P) here and then are sent to
an analog fax machine attached to a T1 (also on the TE410P)
channelbank (CAC1).
Problem is that almost all faxes we send out and receive are mangled...
either only halve pages or very stretched text etc.
Setup in extensions.conf is just:
exten = ${NN_FAX},1,Answer
exten = ${NN_FAX},2,Dial(Zap/49,80)
exten = ${NN_FAX},3,Hangup
echocancel is off for Zap/49 since the path is TDM only
Any pointers to where to look??
Thanks, Michiel
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RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
Adam, Does Polycom license the SIP stuff from Cisco? If not, then Asterisk may be the culprit, because all of my Polycom IP500s exhibit the same behavior. I'm running asterisk 0.7.1, Zaptel CVS and libpri CVS both from a few days ago, but I don't recall having this problem a few months ago when I was running older versions. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Low, Adam Sent: Wednesday, March 10, 2004 1:03 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Well I just took a look at the TAC case and things dont look good, seems the TAC are now blaming Asterisk for the problem but I will go through there debugs and push back, will let you know. -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 08 March 2004 22:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
What versions of Zaptel, Asterisk, and libpri? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Fraizer Sent: Wednesday, March 10, 2004 2:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. For what it's worth, I don't have any delay between answer and audio with my asterisk server and 7960G either originating or answering. It doesn't matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's pretty much instant (not detectable by humans at least). So, there may be some truth to the fact that the delay is caused by the Asterisk install in your case. There are so many variables that it is very hard to tell but, since I don't see the delay, I am leaning towards it being an Asterisk implementation issue. Here's what I'm running: Compaq DL380 1Gha with 1GB of memory Redhat Linux 8.0 (soon to be Gentoo - amazing difference in performance) Asterisk version: CVS-02/15/04-14:03:51 7960 Firmware Version: Application Load ID = P0S3-06-1-00 Boot Load ID = PC030301 DSP Load ID = PS03AT38 I'm using the ULAW codec. John Low, Adam wrote: Well I just took a look at the TAC case and things dont look good, seems the TAC are now blaming Asterisk for the problem but I will go through there debugs and push back, will let you know. -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 08 March 2004 22:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. SIP phone dials 555-1234 (outside line via PRI) 555-1234 rings 555-1234 answers and says Hello SIP phone hears o or nothing at all. If 555-1234 is slow to say something, then everything is heard fine. Caveats. echotraining and echocancel are enabled on the PRI, however, similiar Zap calls are not affected. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Wednesday, March 03, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. When calling out on a Cisco 7960 there is a short delay before the call gets setup and the other side can hear your voice. Anyone know how to compensate for this effect? Sounds like the 7960 has not been configured with a dialplan that supports your * dialplan. Look for the dialplan.xml file on your tftp server and check its contents. Should look something like the following: DIALTEMPLATE TEMPLATE MATCH=0 Timeout=1 User=Phone/ !-- Local operator-- TEMPLATE MATCH=911 Timeout=0 User=Phone/ !-- Local numbers-- TEMPLATE MATCH=3... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=4,4.. Timeout=0 User=Phone/ !-- Local numbers-- TEMPLATE MATCH=5,4.. Timeout=0 User=Phone/ !-- Local numbers-- /DIALTEMPLATE The first entry, above, says if the user dialed 0, then wait for one second to ensure they didn't dial something like 0-555-1212. If no other digits dialed, the 7960 is supposed to send 0 to asterisk after that 1-second timeout. The third entry says my local * extensions are four-digit numbers starting with a 3. If the user dial 3111, the 7960 should immediately send that to * (no timeout). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP image (off-topic)
Buy SmartNet support for the phone. That grants you access to software images through their website. Try Insight. 1-800-INSIGHT. They sell all quantities. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hermann Wecke Sent: Thursday, February 19, 2004 10:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP image (off-topic) My Cisco 7960 is working well with * using SCCP, but I want to change it to SIP. Can anyone here help me on how/where I can buy a SIP image? I contacted a few Cisco partners in the US and some replied will not sell 1 copy/can't handle a small contract and others ignored me. Thanks, Hermann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pick on Cisco 7960's
Title: Message Works fine here. Post your SIP and Zapata configs -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of B. J. BomarSent: Wednesday, February 18, 2004 4:31 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Call Pick on Cisco 7960's Has anyone got the call pick to work on the Cisco 7960's? I have tried to get it to work a couple of time, but all I get is the following error. NOTICE[1142135600]: chan_sip.c:5355 handle_request: Nothing to pick up Thanks, B. J.
RE: [Asterisk-Users] T1 Help
Make sure you have your extensions.conf setup to dial out the T-1. Something like
this.
exten = _81NXXNXX,1,Dial(${LONGDISTANCET1}/${EXTEN:1})
exten = _81NXXNXX,2,Hangup
exten = _71NXXNXX,1,Dial(${LONGDISTANCET1}/${EXTEN:1}||d)
exten = _71NXXNXX,2,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Tuesday, February 17, 2004 1:52 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] T1 Help
I have a question. We have been using Asterisk for a few months with
POT's lines. And have just implemented a T1 Circuit. My problem is I
can receive inbound calls but can't make any outbound calls. We have
Cisco 7940G phones. You will find my config below - if you can find
anything I am doing wrong please let me know.
-gcc
Zapata.conf
[channels]
context=default
group=1
signalling=featd
musiconhold=default
immediate=no
channel = 1-6
zaptel.conf
loadzone=us
defaultzone=us
span=1,0,0,esf,b8zs
em=1-6
sip.conf
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
;externip = 200.201.202.203 ; Address that we're going to put in SIP
messages if we're behind a NAT
;localnet = 192.168.1.0 ; Internal NETWORK address
;localmask = 255.255.255.0 ; Internal netmask
context=default ; Default for incoming calls
;srvlookup = yes; Enable SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for
Pingtel
;tos=lowdelay
;tos=184
;maxexpirey=3600; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;videosupport=yes ; Turn on support for SIP video
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=ilbc
allow=alaw
[tgreene]
type=friend
username=tgreene
fromuser=Todd Greene
secret=dickslap
host=dynamic
canreinvite=no
mailbox=3001
[rnewton]
type=friend
username=rnewton
fromuser=Randy Newton
secret=dickslap
host=dynamic
canreinvite=no
mailbox=3002
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[Asterisk-Users] Wierd Zap Channel Behavior
Here's a wierd one. I'm have a problem where periodically a couple of my extensions dont' get hungup properly. The channel bank doesn't show the channel as active, show channels doesn't show the channel as active, but a zap show channel has the Actual Confinfo: as an active call. This results in the channel receiving one-way audio from an active conversation on another Zap channel. I'm running: Zaptel CVS 2-10-04 (for bigzaplock fix) libpri 0.5.1 asterisk 0.7.1 This happened with zaptel 0.8.1 as well. My guess is that asterisk isn't properly closing the channel when it's hungup. Has anyone seen this behavior? Here's the output of zap show channel 37 File Descriptor: 111 Span: 2 Extension: Context: longdistance Caller ID string: Bad Extension 5239 Destroy: 0 Signalling Type: FXO Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/161, Mode/0x0009 Actual Confmute: No ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] System freeze
Did you possibly have astman running on the localhost? I found that I was getting kernel panics while using astman on an SMP machine with dual T400P cards. Did you see the message on the console before you reset the box? Did you possibly have a serial console connected logging console message? -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jonathan Biggs Sent: Monday, February 09, 2004 1:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] System freeze Currently in progress of trying to debug similar problem on my own system. Sometimes it happened during call transfers, but this last time, it happened all by itself at 4:00 AM, no calls even close. Complete system Freeze, Nothing at all workings, except the reset button. You setup is vastly different from mine to. Dual Pentium III SMP, X100P Dual TDM400P What type and version of Linux? Mine is RH9 2.4.20-8??? Would love to track this one down... --- Michael Welter [EMAIL PROTECTED] wrote: I have a Gigabyte K7 motherboard with an Athlon 2400+ processor. Before the T1 install I had two T100P cards, one for the channel bank and the other unused. This ran perfect for a month. Last week we installed a new integrated T1 into the unused T100P (to replace POTS lines and DSL.) In BIOS, I disabled some unused peripherals so that each T100P would find its own unique IRQ. I also installed the updated asterisk, libpri, and zaptel sources. I have seen two system freezes--one on Friday and one this morning. The whole system freezes--no LAN, no phones, no console. During this morning's freeze there were no calls in progress. The logs say nothing. Has anyone else seen this? I suspect it isn't an asterisk problem, but I would appreciate feedback. Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial-out and Dial-in modem problems.
Has anyone experienced problems with dialup through asterisk. I'm having some varied success with dial-in and dial-out. All my analog extensions are connected to * via Adtran 750 FXS channelbanks using FXO_KS signalling. I have a longdistance T-1 (em_w) from sprint and a local T-1 PRI from Verizon. I'm running Asterisk 0.7.1, Zaptel 0.8.0, and libpri 0.5.1. The problem that I'm seeing is that active modem connections (in or out) are hanging up randomly. I have busydetect=yes and busycount=6 on all my non-PRI zap channels. I have echocancel=128 and echocancelwhenbridged=yes on all of my zap channels. Tonite, I plan on uncommenting BUSYDETECT+= -DBUSYDETECT_COMPARE_TONE_AND_SILENCE and CFLAGS+=-DOLD_DSP_ROUTINES in the asterisk Makefile to see if this resolves the issue. Has anyone had any experience with dialup problems like this. Regards, -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 quick dial
Take a look at dialplan.xml on your tftp server. DIALTEMPLATE TEMPLATE MATCH=0 Timeout=1 User=IP/ !-- Local operator-- TEMPLATE MATCH=8,011* Timeout=6 User=IP/ !-- International calls-- TEMPLATE MATCH=8,1.. Timeout=0 User=IP/ !-- Long Distance -- TEMPLATE MATCH=9,1.. Timeout=0 User=IP/ !-- Toll Free -- TEMPLATE MATCH=9,11 Timeout=0 User=IP Route=Emergency Rewrite=9911/ TEMPLATE MATCH=9,.. Timeout=0 User=IP/ !-- Local numbers -- TEMPLATE MATCH=9,.11 Timeout=0 User=IP/ !-- Service numbers -- TEMPLATE MATCH=78.. Timeout=1 User=IP/ !-- Corporate Dial plan-- TEMPLATE MATCH=52.. Timeout=1 User=IP/ !-- Corporate Dial plan-- TEMPLATE MATCH=87.. Timeout=1 User=IP/ !-- Corporate Dial plan-- TEMPLATE MATCH=5000 Timeout=1 User=IP/ !-- Voicemail -- TEMPLATE MATCH=4... Timeout=1 User=IP/ !-- Meetme -- TEMPLATE MATCH=11.. Timeout=1 User=IP/ !-- Parking -- TEMPLATE MATCH=* Timeout=15/ !-- Anything else -- TEMPLATE MATCH=123#45#6 Timeout=0 User=IP/ !-- Match `#' -- TEMPLATE MATCH=12\*345Timeout=0 User=IP/ !-- Match * Char -- /DIALTEMPLATE -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jose Inzunza/YM/RWDOE Sent: Tuesday, February 03, 2004 11:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 quick dial Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the call would initiate once I pressed the '3'. Any help would be appreciated. Jose ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran 750 DID question.
Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 EM wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup EM in zaptel.conf and EM_W in zapata.conf. They work, however, no DNIS info is being passed. Do I need to signal these something different like loopstart or kewlstart, so the DNIS info gets passed? I watch the Tx/Rx bits from zttool, and everything looks okay coming from the Adtran. It looks like asterisk isn't winking properly. When I had the lines misconfigured for fxs_ls the DNIS info was passing fine. I'm running zaptel-0.8.0 libpri-0.5.1 And asterisk CVS from 12/23/2003 RedHat 8.0 Dual 2.4 Xeon Processors (hyperthreading disabled) 2Gig Memory Any help would be greatly appreciated. Regards, -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adtran 750 DID question.
I tried both featd and em in zapata.conf, to no avail. I restarted in between all changes. Is it possible to signal the DPO ports on the 750 with fxo_ls or fxo_ks? This is the last piece to my DID puzzle. Anyone else with experience on this oddball config? Thanks, -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Sharp Sent: Friday, January 30, 2004 11:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Adtran 750 DID question. Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 EM wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup EM in zaptel.conf and EM_W in zapata.conf. They work, however, no DNIS info is being passed. Do I need to signal these something different like loopstart or kewlstart, so the DNIS info gets passed? I watch the Tx/Rx bits from zttool, and everything looks okay coming from the Adtran. It looks like asterisk isn't winking properly. I had a similar problem. I ended up setting the trunks to either just plain em or featd (I don't remember). I chased through the chan_zap source code and decided (maybe incorrectly) that asterisk doesn't look for DNIS digits in EM Wink mode. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adtran 750 DID question.
Yes. Adtran FXS cards. Did you say you were using Adtran FXS cards? Bisker, Scott (7805) wrote: Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 EM wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup EM in zaptel.conf and EM_W in zapata.conf. They work, however, no DNIS info is being passed. Do I need to signal these something different like loopstart or kewlstart, so the DNIS info gets passed? I watch the Tx/Rx bits from zttool, and everything looks okay coming from the Adtran. It looks like asterisk isn't winking properly. When I had the lines misconfigured for fxs_ls the DNIS info was passing fine. I'm running zaptel-0.8.0 libpri-0.5.1 And asterisk CVS from 12/23/2003 RedHat 8.0 Dual 2.4 Xeon Processors (hyperthreading disabled) 2Gig Memory Any help would be greatly appreciated. Regards, -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling zaptel
I take it you are running RedHat 8 (or 9) since this is the most up-to-date kernel. Did you install the kernel-sources and kernel-util rpms as well? You'll need these in order to compile and install zaptel. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of T. Chan Sent: Friday, January 30, 2004 4:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling zaptel Dear all, I have been testing with Asterisk for a bit of time and yesterday I tried to upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not able to compile Zaptel. The kernel runs good and everything intact, I was trying to recompile Asterisk in order to make sure that everything was clean. I have gone into /usr/src/zaptel, done a make clean and then done a make install as what I have always done after updating the asterisk version. However, now I am getting the following error, wct4xxp.c: In function 't4-interrupt' wct4xxp.c: 1357:structure has no member named 'Lock' make: *** [wct4xxp.o] error and then it stopped compiling, can someone please let me know if I am missing something please, greatly appreciated. thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Adtran 750 DID question.
I guess asterisk is winking properly then, because the line rings when dialed. In zaptel.conf the lines are set to em and in zapata.conf they are set to em_w. The FXS cards are series L1. What I'm seeing is that the DNIS info is not being passed through to asterisk. Since I get no DNIS, it shoves the call to my s extension. This is what I see on console -- Starting simple switch on 'Zap/3-1' Once I see the call come in I do a show channel and here is what I get. -- General -- Name: Zap/3-1 Type: Zap UniqueID: 1075499367.29 Caller ID: (N/A) DNID Digits: (N/A) State: Ring (4) Rings: 1 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor: 48 Frames in: 212 Frames out: 0 Time to Hangup: 0 -- PBX -- Context: local Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (None) Stack: -1 Blocking in: ast_waitfor_nandfds Any ideas on this? -sb ... When some one calls into your DID Trunk line what symptom do you see on Adtran as well as on Asterisks console? Asterisk winks properly in FXO DPO mode, we had checked this things with our Telco instruments. If your PBX doesn't wink to Telco, your DID line's will become busy, that is one way to check whether Asterisk is winking properly or not. What series FXS card you are using? Is it L1 or L2? Because only L1 series cards work with our DID trunk line. I don't know why and still Adtran technical support is not able to figure out. I have same configuration running in my office and finally with all the help from Digium and adtran, problem seems to be less. Still fully it is not resolved yet because FXS L2 is not working. Hopefully Adtran will release new firmware. Regards, Kekin -Original Message- From: Kekin Dand Sent: Tuesday, January 27, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: Re:Incoming DID call Voice Problems I had similar problem and it took all most 2 months to resolve it. Few things you have check in your Adtran 750 configuration. 1. For incoming DID trunk line it has to terminated on FXS card. (Which I think you already did) 2. FXS card needs to be set on FXS DPO mode in order to work properly. 3. Your DID trunk line should be configured in Asterisk as em in zaptel.conf and em_w in Zapata.conf. Reason you are facing this problem either your battery is not getting reversed and Telco can't see Answer Supervision on your line when the calls get connected. If you have ohms meter or multimeter check your DID line voltages. When inbound call comes in and both parties go off hook you should see positive voltage on that line, in idle situation(on hook) you will negative 48V. Yes, local calls are different then long distance. Answer Supervision is not required for local calls. Only for long distance it is required, so that Telco can start billing. Hope this should resolve your problem. Regards, Kekin Subject: RE: [Asterisk-Users] Incoming DID call Voice Problems Date: Mon, 26 Jan 2004 09:31:32 -0500 From: Bisker, Scott (7805) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] I have an updated question on this one. It seems that only inbound long = distance calls (calls from outside the local calling area) on our DID = trunk have one-way voice. I have my adtran 750 fxs lines configured as = FXS Loopstart with all the defaults. Again, the problem is that once = the call bridges, the outside caller can hear the person they called, = but the inside person can't hear the caller. This happens regardless of = the internal technology, SIP, Zap, H323. Could it be possible that inbound long distance calls are signalled = different than inbound local calls? Inbound calls on the PRI work = flawlessly. Any ideas -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Adtran 750 DID question.
Yes. immediate=no is in zapata.conf before the channel declaration. This makes absolutely no sense at all. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Don Pobanz Sent: Friday, January 30, 2004 5:11 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Re: Adtran 750 DID question. On Friday, January 30, 2004 3:56 PM, Bisker, Scott (7805) [SMTP:[EMAIL PROTECTED] wrote: I guess asterisk is winking properly then, because the line rings when dialed. In zaptel.conf the lines are set to em and in zapata.conf they are set to em_w. The FXS cards are series L1. What I'm seeing is that the DNIS info is not being passed through to asterisk. Since I get no DNIS, it shoves the call to my s extension. Have you verified that immediate = no in zapata.conf? If not, then * may not be waiting for the digits before trying to find a match. Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming DID call Voice Problems
I have an updated question on this one. It seems that only inbound long distance calls (calls from outside the local calling area) on our DID trunk have one-way voice. I have my adtran 750 fxs lines configured as FXS Loopstart with all the defaults. Again, the problem is that once the call bridges, the outside caller can hear the person they called, but the inside person can't hear the caller. This happens regardless of the internal technology, SIP, Zap, H323. Could it be possible that inbound long distance calls are signalled different than inbound local calls? Inbound calls on the PRI work flawlessly. Any ideas -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Saturday, January 24, 2004 3:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Incoming DID call Voice Problems Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but their own echo. Calling party hears everything fine. I have MARK2 enabled in Zaptel driver for echo problems on my PRI line. I can't seem to replicate the problem calling out PRI to the DIDs, or from a cell phone. I can reliably replicate the problem with an offsite customer that calls in. Any idea what may be causing this? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming DID call Voice Problems
Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but their own echo. Calling party hears everything fine. I have MARK2 enabled in Zaptel driver for echo problems on my PRI line. I can't seem to replicate the problem calling out PRI to the DIDs, or from a cell phone. I can reliably replicate the problem with an offsite customer that calls in. Any idea what may be causing this? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] B-channels restart problem
Ali,
If
Zap/82 is channel 20 on Span 3, then it looks like it's hanging up before the
channel restarts as this line indicates.
== Spawn extension (inbound, 9009170,
2) exited non-zero on 'Zap/82-1'
Maybe
there is a problem with your agi script.
B
channels only restart when the PRI line isn't busy.
-sb
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Ali
MughrabiSent: Thursday, January 15, 2004 7:48 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] B-channels
restart problem
Hi ,
I'm having a problem that really bothers me , I havelooked
for similar cases but couldn't really
find an answer .
I keep getting messages whichsays that
-- B-channel xx successfully restarted on span x
and this causes the calls to be disconnected if somone is already using
the channel that is being restarted, plz notice below the under lined
textin which I was testing and got on channel 20 span 3 , and then
it got disconnected after restart, I can't decide if this is a telco or
configuration problem , telco says they have no problem , shall I beleive
them?
please I need any help or comment that might be helpful.
Thanx in Advance
pleaes reply here or toat [EMAIL PROTECTED]
Thanx in Advance
Ali Mughrabi
-- Accepting call from '065639815' to '9009170' on channel 20, span 3
-- Executing AGI("Zap/82-1",
"../album_show/album_show.agi|--apelant=065639815") in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/../album_show/album_show.agi
-- B-channel 14 successfully restarted on span 3
-- B-channel 15 successfully restarted on span 3
== Spawn extension (inbound, 9009170, 2) exited non-zero on
'Zap/82-1'
-- Hungup 'Zap/82-1'
-- B-channel 17 successfully restarted on span 3
-- B-channel 18 successfully restarted on span 3
-- B-channel 20 restarted on span 3
-- B-channel 19 successfully restarted on span 3
-- B-channel 20 successfully restarted on span 3
-- B-channel 21 successfully restarted on span 3
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RE: [Asterisk-Users] Auto Starting Asterisk
An even better way to get asterisk started is to use the init scripts provided with asterisk and the zaptel kernel modules. cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel Then do the proper linking, etc to get asterisk to start in your current run level. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David J Carter Sent: Tuesday, December 23, 2003 7:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk Hi, In rc.local I added the line /etc/rc.d/run-asterisk I then created a small script of 2 lines called run-asterisk #!/bin/sh /usr/sbin/asterisk do a chmod 755 on the file and reboot. The Asterisk server then starts at every reboot. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adthrawn Sent: 23 December 2003 12:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Auto Starting Asterisk Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Readline readline-devel installation on RH9
Ariel, You can install them from the RH9 CD. Also, make sure you use readline and not redline. Insert the RH9 CD cd /mnt/cdrom/RedHat/RPMS rpm -ivh readline*.rpm You may need to switch CDs in order to find the correct disc. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, December 17, 2003 1:19 PM To: Asterisk User List Subject: [Asterisk-Users] Readline readline-devel installation on RH9 I have a new user question. Sorry I know most of you are Linux experts I am not! I am just getting my feet wet with this. And I am sorry to ask this stupid question. I was following an installation post from Wiki that said when using RH 9 you need to make sure that you have the following installed first and you should check them with the following command. Are there any other items I need to check on. I have been having problems setting up asterisk so I want to rule out the OS first. # rpm -q kernel-source redline redline-devel openssl opessl-devel I have done this but my system reports that redline and readline-devel not installed. How do I install these items without re-installing RH 9 all over again? Also why are these needed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patch to fix vmail.cgi forwarding problem
Hello All,
Here is a patch that fixes the problem when forwarding messages with vmail.cgi. Bug
submitted with patch on bugs.digium.com.
-sb
--- /usr/src/asterisk/vmail.cgi.orig2003-12-17 14:21:47.0 -0500
+++ /usr/src/asterisk/vmail.cgi 2003-12-17 15:07:36.0 -0500
@@ -672,7 +672,7 @@
sub message_copy()
{
- my ($mbox, $oldfolder, $old, $newmbox, $new) = @_;
+ my ($mbox, $newmbox, $oldfolder, $old, $new) = @_;
my $oldfile, $newfile;
return if ($mbox eq $newmbox);
@@ -788,7 +788,7 @@
# print header;
foreach $msg (@msgs) {
# print Forwarding $msg from $mbox to $newmboxBR\n;
- message_copy($context, $mbox, $folder, $msg, $newmbox,
sprintf %04d, $msgcount);
+ message_copy($mbox, $newmbox, $folder, $msg, sprintf %04d,
$msgcount);
$msgcount++;
}
$txt = Forwarded messages . join(', ', @msgs) . to $newmbox;
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[Asterisk-Users] IP 500/600 1.1.0 Firmware
Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: estara softphone problem
Could you post the console output from when you run the softphone application? Maybe there is a problem with registration. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong Sent: Friday, December 12, 2003 5:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: estara softphone problem Hi Scott, my 7960s can call each other without any problem. I changed ip.conf as you recommended, and still didn't work. But from sip show peers, it looks like my softphone is not talking to asterisk properly. Asterisk got the softphone's IP address, but its status is unreachable, I'm trying to figure out why. Line-8001 till Line-8004 are Cisco 7960s (two phones, each with two lines defined), they seem to be OK. Monet*CLI sip show peers Name/usernameHost Mask Port Status 8005/800510.26.6.78 (D) 255.255.255.255 5060 UNREACHABLE Line-8004/Line- 10.26.6.198 (D) 255.255.255.255 5060 OK (51 ms) Line-8003/Line- 10.26.6.198 (D) 255.255.255.255 5060 OK (51 ms) Line-8002/Line- 10.26.6.129 (D) 255.255.255.255 5060 OK (41 ms) Line-8001/Line- 10.26.6.129 (D) 255.255.255.255 5060 OK (41 ms) Here is the console message when I made a call from the softphone to the Cisco 7960 (which was successful). -- Executing Macro(SIP/8005-4971, stdexten|8001|SIP/Line-8001) in new stack -- Executing Dial(SIP/8005-4971, SIP/Line-8001|20|t) in new stack -- Called Line-8001 -- SIP/Line-8001-f19d is ringing == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/8005-4971' in macro 'stdexten' == Spawn extension (default, s, 1) exited non-zero on 'SIP/8005-4971' And here is the console message when I tried to call the softphone from Cisco 7960 (which failed). -- Executing Dial(SIP/Line-8001-d450, SIP/8005) in new stack == Everyone is busy at this time -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott Sent: Friday, December 12, 2003 3:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] estara softphone problem __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] estara softphone problem
In sip.conf do you have type=friend for your softphone? If not you'll only be able to send or receive calls depending on the option you selected. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong Sent: Friday, December 12, 2003 2:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] estara softphone problem Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying unable to create channel to my softphone. What could be the problem? I searched the archive with no luck. When you reply, please copy to [EMAIL PROTECTED], really appreciate it! __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: estara softphone problem
I had a similar problem with my 7960 phones. It ended up being a problem with quotes in the SIP.cnf file. Do a sip show peers from the console to see if the 7960 is registered properly. For a test set the following values in the cnf file line1_name: 8005 line1_shortname: 8005 line1_authname: 8005 line1_password: 8005 line1_displayname: 8005 In sip.conf change the entry to [8005] type=friend username=8005 secret=8005 canreinvite=no host=dynamic mailbox=8005 callerid=Hao Zhong Desk8005 nat=no Reload asterisk and reboot the phone. This should get you up and running. I know it's changing everything, but get it working with this config, then change one var at a time until you find the one that is causing you problems. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong Sent: Friday, December 12, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: estara softphone problem Hi Scott, thanks for the reply. Here is how my sip.conf looks like for the softphone, I tried type '3Dfriend' and asterisk didn't like it. [hzhong-desk] type=friend username=hzhong-desk callerid=Hao Zhong Desk 8005 mailbox=8005 secret=cisco nat=no host=dynamic canreinvite=no qualify=200 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott Sent: Friday, December 12, 2003 2:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] estara softphone problem In sip.conf do you have=20 type=3Dfriend for your softphone? If not you'll only be able to send or receive calls depending on the = option you selected. -sb __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 6.0 + Asterisk question
John, I have 12 7960 phones with 6.0 with no issues. Sounds like a hardware problem to me. -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Sunday, November 30, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 6.0 + Asterisk question I have several phones running Cisco's 6.0 SIP software release at this time. Two of the phones have not shown any abnormal behaviors, but one of them has an unsettling propensity to lock up after several hours, where the softkey labels disappear and the phone stops registering, requiring the standard *-6-settings reboot sequence. Otherwise, the phone seems to work OK except for a slight flickering of the LCD (hence my suspicions that this might be a hardware issue.) The two working 6.0 phones I have are registered to Asterisk CVS-11/08/03-20:12:44 and the one failing phone is registering to Asterisk CVS-11/18/03-16:53:17. I can't easily move phones around at the moment due to a variety of infrastructure and political issues beyond my control, so I ask here if anyone else here has experienced any unexpected lockups with 6.0 (registering to Asterisk or not) or if this is a hardware problem with this particular phone. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Issue
Michael, Where in your extension definition to you dial a channel (SIP, Zap, or other)? You are missing the dial entry. -sb -Original Message- From: Lists [mailto:[EMAIL PROTECTED] Sent: Saturday, November 29, 2003 10:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sip Issue Hi all I am having some issues with a gs 100 phone. It is on the same network as my * server. There is no firewall. In extentions.conf exten = 5,1,Answer exten = 5,2,MusicOnHold(default) When I dial 5 from the sip phone -- Executing Answer(SIP/mlh-2e75, ) in new stack -- Executing MusicOnHold(SIP/mlh-2e75, default) in new stack -- Started music on hold, class 'default', on SIP/mlh-2e75 ---about 7 secs... -- Stopped music on hold on SIP/mlh-2e75 == Spawn extension (sip, 5, 2) exited non-zero on 'SIP/mlh-2e75' In /var/log/asterisk/messages Nov 29 23:01:46 WARNING[1142127920]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 28503 (Response) Any Ideas? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco 7960 power suplies?
You can get them from any cisco reseller. If you are in the US, the part number is CP-PWR-CUBE= -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lists Sent: Sunday, November 30, 2003 6:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco 7960 power suplies? Does anyone know where to get cisco 7960 power suplies? What should they cost? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] double-dial in SIP Grandstream
Marc, This is the typical behavior for call waiting. While you are initiating a call, people who call your number will get a busy signal until your first call connects. Once the call connects, the number 2 caller will hear a ring until you pickup. If you want to disable callwaiting then put callwaiting=no in sip.conf for that particular alias. [alias] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Overhead Paging
Title: Message Jerry, Do you have it setup so that multiple phones answer one extension? I tried that setup with two Cisco phones, however, only the quickest responding phone answered. If you have a config that rings multiple phones and all of the phones answer the same call, I'd be interested to see the config. I guess theway to do it would be to setup a meetme conference and then dial all parties into the conference then speak -sb -Original Message-From: Jerry Gibson [mailto:[EMAIL PROTECTED]Sent: Friday, November 14, 2003 8:52 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Overhead Paging We do the same thing with the Snom phones. They can be set up for auto-answer, and they have a speaker jack in the back that is the same levels as a sound card on a PC. And the Snom phone automaticly hangs up when the caller hang up is detected (the SIP BYE message). Jerry -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805)Sent: Thursday, November 13, 2003 6:17 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Overhead Paging Our setup is to set the OSS device to autoanswer. The output of the soundcard feeds into a bank of overhead speakers. If the channel is in use, then the call gets put in a queue until the OSS device is free. -sb -Original Message-From: Johnson, Randy [mailto:[EMAIL PROTECTED]Sent: Thursday, November 13, 2003 5:34 PMTo: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Overhead Paging Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson
RE: [Asterisk-Users] Overhead Paging
Title: Overhead Paging Our setup is to set the OSS device to autoanswer. The output of the soundcard feeds into a bank of overhead speakers. If the channel is in use, then the call gets put in a queue until the OSS device is free. -sb -Original Message-From: Johnson, Randy [mailto:[EMAIL PROTECTED]Sent: Thursday, November 13, 2003 5:34 PMTo: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Overhead Paging Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson
RE: [Asterisk-Users] Red Alarm
How far is your server from the telco box? I found that with extended distances, my reliabilty was significantly decreased. If you still have problems, check your RJ-48X jack for connection problems. -sb -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 5:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Red Alarm On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz [EMAIL PROTECTED] wrote: Sometimes I receive a Red Alarm in my E1 trunk (EM immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? I'd suggest your telco doing loopup and checking the circuit. My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Soundpoint IP600
Default User Password is 123 Default Admin Password is 456 -sb -Original Message- From: Roman Pelikh [mailto:[EMAIL PROTECTED] Sent: Friday, October 31, 2003 11:54 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Polycom Soundpoint IP600 Does anyone have the Admin password for the phone in order to change configuration Roman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 500
Title: Polycom SoundPoint IP 500 The SIP version of the IP500 runs the same firmware, etc as the IP600. The config files are the same. The only difference is that the IP500 has three lines instead of six. I believe that the model number is the same for all IP500 phones, its just the firmware that's different. But, like Matt said, unless you have a copy of the working firmware, I wouldn't try it unless you are willing to potentially render the phone useless in case of an incompatability. -sb -Original Message-From: mattf [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 29, 2003 9:52 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 500 Hello, I only have experience with the IP600 which is SIP only, IP500 is supposedly capable of SIP but you would need to get the firmware from Polycom. I am in the process of trying to sign up for their developer program, but it is a SLOOOW process. I do have the firmware for the IP600 but it is anyone's guess that it would work with the IP500 and I wouldn't want you to ruin your phone trying. The IP600 is a great phone with lots of great features and a good design. Let us know if you get it working. I'll let you know if I get a copy of the IP500 SIP firmware. MATT--- -Original Message-From: Ed Rubright [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 29, 2003 7:52 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Polycom SoundPoint IP 500 Hello all, Has anyone used the SIP version of this phone with Asterisk? I see Polycom has a H.323 and MGCP version also, does anyone know if you flash the phone to swith protocols? Thanks in advance for the info. Ed
RE: [Asterisk-Users] Newbie hardware question
I have 6 750s attached to my pbx server. The 850s have a lot of functionality you don't really need. -sb -Original Message- From: TC [mailto:[EMAIL PROTECTED] Sent: Thursday, October 30, 2003 1:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie hardware question You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. hmm what adtran unit is that the most popular adtran cb's used with * are the ta-750/850 and the slots are provisioned with 4 channels per slot/card total 6 slots per unit, 24 channels total ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QOS
Pretty much anything from Cisco or Foundry support QOS. Linux and BSD support it as well. -sb -Original Message- From: Nick Knight [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 28, 2003 6:52 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] QOS Hello all, Apologies as not really an Asterisk question - QOS. I have been told to implement VOIP correctly you need QOS implemented across the network as a whole. What network switches support this? Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM 400P signal problem
Jim, What type of cabling are you using? What's terminated on the other end of each port (Channel Bank, Telco Demarc?) How far away are you from what's connected on cards 2 3? This will have a lot to do with signal and noise? -sb -Original Message- From: Jim Paraschou [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 28, 2003 4:34 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TDM 400P signal problem Hi everybody, I have 3 TDM400P installed in a machine,and though the 4 ports of the first card work fine, some ports on the other two have low or no signal and a noise instead. Can someone help? Thanx __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: # [Asterisk-Users] TDM 400P signal problem
my mistake, I was thinking a T-1 card. -sb -Original Message- From: Jim Paraschou [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 28, 2003 4:57 PM To: [EMAIL PROTECTED] Subject: # [Asterisk-Users] TDM 400P signal problem It is a cable 4-5 meters long that has handssets connected I don't think its a matter of a distance __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup (*8) on SIP devices.
Just submitted a patch for this on asterisk-dev.
Quick fix add the following line above line 5022 in chan_sip.c
ast_setstate(c,AST_STATE_DOWN);
Should look like this when you are done.
} else {
5021ast_mutex_unlock(p-lock);
5022ast_setstate(c,
AST_STATE_DOWN);
5023ast_hangup(c);
5024ast_mutex_lock(p-lock);
c = NULL;
-Scott
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 2:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Yes
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
Thank's.
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[Asterisk-Users] WAS: Call pickup (*8) on SIP devices. Bug #116
I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(13) are Cisco 7960's and SIP(2) is a Polycom IP500 -- I've also tried with SIP(2) being a 7960 as well. In scenario 2, when SIP(1) hangs up, a CANCEL message is sent to SIP(2). In scenario 1, when SIP(3) picks up the call to SIP(2), SIP(2) never receives a CANCEL message, thus, it continues to ring. At the end of the debug, after SIP(2) stop's ringing, it sends 3 Decline messages to the asterisk PBX. If you need any more debug info, let me know. -sb *CLI sip debug SIP Debugging Enabled Sip read: INVITE sip:[EMAIL PROTECTED];user=ip SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060 From: 5285 sip:[EMAIL PROTECTED];tag=000d287e269a000f5181f06d-45b64a55 To: sip:[EMAIL PROTECTED];user=ip Call-ID: [EMAIL PROTECTED] Date: Thu, 23 Oct 2003 21:23:19 GMT CSeq: 101 INVITE User-Agent: CSCO/5 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 246 Accept: application/sdp Remote-Party-ID: 5285 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 9972 27311 IN IP4 192.168.1.84 s=SIP Call c=IN IP4 192.168.1.84 t=0 0 m=audio 31790 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 192.168.1.84 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.84:5060 From: 5285 sip:[EMAIL PROTECTED];tag=000d287e269a000f5181f06d-45b64a55 To: sip:[EMAIL PROTECTED];user=ip;tag=as4284ac7e Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=676c94f0 Content-Length: 0 to 192.168.1.84:5060 Sip read: ACK sip:[EMAIL PROTECTED];user=ip SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060 From: 5285 sip:[EMAIL PROTECTED];tag=000d287e269a000f5181f06d-45b64a55 To: sip:[EMAIL PROTECTED];user=ip;tag=as4284ac7e Call-ID: [EMAIL PROTECTED] Date: Thu, 23 Oct 2003 21:23:19 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED];user=ip SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060 From: 5285 sip:[EMAIL PROTECTED];tag=000d287e269a000f5181f06d-45b64a55 To: sip:[EMAIL PROTECTED];user=ip Call-ID: [EMAIL PROTECTED] Date: Thu, 23 Oct 2003 21:23:19 GMT CSeq: 102 INVITE User-Agent: CSCO/5 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=5285,realm=asterisk,uri=sip:192.168.1.15,response=5025d36a5940ca107c7bdce5aa 1b7e99,nonce=676c94f0,algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 246 Remote-Party-ID: 5285 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 9972 27311 IN IP4 192.168.1.84 s=SIP Call c=IN IP4 192.168.1.84 t=0 0 m=audio 31790 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 192.168.1.84 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 8719 in default list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.84:5060 From: 5285 sip:[EMAIL PROTECTED];tag=000d287e269a000f5181f06d-45b64a55 To: sip:[EMAIL PROTECTED];user=ip;tag=as710b2362 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.84:5060 -- Executing Macro(SIP/5285-6f79, stdexten|8719|SIP/test1) in new stack -- Executing DBget(SIP/5285-6f79, temp=CS/8719) in new stack -- DBget: varname=temp, family=CS, key=8719 -- DBget: set variable temp to 0 -- Executing GotoIf(SIP/5285-6f79, 0?s|4) in new stack WARNING[229391]: File pbx.c, Line 4442 (pbx_builtin_gotoif): Not taking any branch -- Executing
RE: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization
I found the best way to upgrade is install Red Carpet from www.ximian.com. Subscribe to the RH 9.0 channel. And do a complete update. The only drawback is that this method doesn't update the kernel. To do the kernel, ftp the latest kernel from updates.redhat.com. rpm -ivh latest kernel.rpm. Change /etc/grub.conf to reflect the newest kernel is the default. Scott -Original Message- From: David Luyens [mailto:[EMAIL PROTECTED] Sent: Monday, September 29, 2003 10:07 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization Hi, what is the best way to upgrade rh 9.0 installed from cd? David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Wednesday, September 24, 2003 8:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization i run some system with * rh 9.0 be sure to have latest updates (install them after installing redhat, and before installing *) and check to have mpg123 installed (you must get it on the mpg123 website), since redhat has a mpg321 replacement that won't work with * matteo. Il mer, 2003-09-24 alle 20:23, James Ray ha scritto: Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323.c compile error
Hello all,
I got the following error compiling h323 support in the latest cvs. Below
the error is a diff to the file that I got to make it work. I took an
example out of sip as far as the syntax for ast_rtp_new. Not sure if it is
correct or not, but it seems to work. Please correct me if I am wrong in
the additional 2 arguements.
Regards,
Scott
cc -g -pg -c -o chan_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes
-Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE
-DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING
-DP_USE_PRAGMA -I/usr/local/pwlib/include/ptlib/unix
-I/usr/local/pwlib/include -I/usr/local/openh323/include
-Wno-missing-prototypes -Wno-missing-declarations chan_h323.c
chan_h323.c: In function `oh323_alloc':
chan_h323.c:687: too few arguments to function `ast_rtp_new'
chan_h323.c: At top level:
chan_h323.c:1601: warning: initialization from incompatible pointer type
make: *** [chan_h323.o] Error 1
--- chan_h323.c 2003-07-01 08:09:33.0 -0400
+++ chan_h323.c.mod 2003-06-30 10:25:30.0 -0400
@@ -684,7 +684,7 @@
/* Keep track of stuff */
memset(p, 0, sizeof(struct oh323_pvt));
- p-rtp = ast_rtp_new(NULL, NULL);
+ p-rtp = ast_rtp_new(NULL, NULL, 1, 0);
if (!p-rtp) {
ast_log(LOG_WARNING, Unable to create RTP session: %s\n,
strerror(errno));
free(p);
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[Asterisk-Users] H.323 CallerID
Hello All, Couple of quick (hopefully) questions. 1. I noticed in the latest h.323 cvs log that callerid is now supported. Is there any special configuration needed to get this to work. I have tried callerid= in h323.conf to no avail. Calls from a h.323 device show callerid as the user e.g., [h323user01] would show as h323user01 and calls to the h.323 device show callerid as root. Any help on this would be greatly appreciated. As always, thanks in advance for any help. Regards, -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Issue with connected TA 750
Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a bum BCU? Also, when performing a ring test from the admin port of the 750, the same behavior is present. Any ideas on this one? Thanks in advance. Scott Bisker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
