Re: [asterisk-users] Unable to execute
Is SELinux enabled on the machine? If it is, you might have a problem with
the asterisk process being able to execute in that directory.
On Wed, Nov 11, 2009 at 4:38 AM, her Garcia herli...@lycos.com wrote:
Hello. I am trying to execute an fax reception script and i am getting the
following:
[Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper:
Unable to execute '/var/lib/asterisk/scripts/mailfax '
I tried changing the permissions to the mailfax script but Asterisk still
can´t execute the script.
The macro is the following:
[macro-faxreceive] exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${CALLEDFAX}/${UNIQUEID}) exten =
s,2,Set(EXTEMAIL=${MACRO_EXTEN}/xEmail) exten = s,3,NoOP() exten =
s,4,Set(EXTNAME=${MACRO_EXTEN}/xName) exten = s,5,NoOP() exten =!
s,6,Set(EXTCOMPANY=${MACRO_EXTEN}/xCompany) exten =
s,7,nvfaxdetect(${FAXFILE}.tif) exten = s,103,Set(EXTMAIL=
hsalvare...@bit2net.com) exten = s,104,Goto(7) exten =
s,105,Set(EXTNAME=Hernan) exten = s,106,Goto(7) exten =
s,107,Set(EXTCOMPANY=Bit2net S.A) exten = s,108,Goto(7)
[fax] exten = 9299,1,Macro(faxreceive)
exten = h,1,System(/var/lib/asterisk/scripts/mailfax ${CALLERIDNUM}
${CALLEDFAX} ${EXTNAME} ${EXTEMAIL} ${FAXFILE} ${EXTCOMPANY})
Any ideas? Thanks in advanceHernán
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Re: [asterisk-users] question on capacity
Jerry Geis wrote: Can one server (like AMD 6000+ X2) with 2 GIG ram running asterisk 1.4 handle having 2100 wireless phones connected. All phones will not be talking at the same time only a couple will be. There may be 1 T1 card in the box. Will this work? If not how does one handle this situation. Any transcoding involved, or will you be using G711u on all the phones? How many, max, do you see talking at the same time? What will the reregistration interval be for the phones (How often will they check into the Asterisk box)? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting a call back from voicemail?
Mike Dent wrote: Hi, is there a way or feature available in Asterisk where one can 'pull' a call back from voicemail. i.e. if you don't get to the phone in time and it goes to voicemail, can you key some sequence in and pull the caller out of voicemail and speak to them? It seems like you should be able to transfer the caller's channel to another extension.. That extension would ring, though, so it wouldn't be an immediate connect. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P
Deepak Naidu wrote: Hi, I am currently using TE110P Digium card on a PRI card. Basically the echo is so much that one can disticntly identify that. I have tried all the combination if tuning configuration seen in forums etc. I am using MG2 cancellor algorithm also tuned the RX TX gains, still there is an echo. So I am thing to purchase an hardware based echo cancellor like Digium Wildcard TE212P. So in this regards I would like to get some view whether its worth to buy a hardwrae based echo cancellor. Will this resolve the issue, or will be just waste of money. I am using Asterisk 1.2.18 latest version of zaptel drivers. Hope if someone had the same issue, I what has done to resolve it would be much appreciable. In my experience, it is well worth the money. After installing several for customers, we never bought the non-HWEC cards again... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Polycom IP 501 is formatted its file systemitself
Crazy Boy wrote: Hi Steven, Thank you for very much for your response. I solved this problem. I found these files in Internet and setup FTP server and uploaded these files into that. Now, my phone is working. Thank you. Great news! You might want to keep those files on hand (burn to CD or something), just in case you need them again. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Polycom IP 501 is formatted its file systemitself
No problem. Sorry about the delay. Unfortunately, I no longer work for a Polycom reseller, so I can't give you a simple link to the files that you need (someone else on the list might be able to help you out off-list with this). Sometimes, you can find them online by searching google with Polycom 501 firmware. Sorry that I can't be of more help. If I come across the image files, I will send you an email and let you know. Best of luck! Steve Crazy Boy wrote: Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. */Steve Totaro [EMAIL PROTECTED]/* wrote: We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Polycom IP 501 is formatted its file system itself
Noah Miller wrote: Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. If you can get to the boot menu (where it offers to let you configure a server to boot off of), you can recover with a firmware image. You usually can get these only from resellers (because Polycom doesn't want to deal with customer support on an individual basis). Let me know if you can get this far; I might be able to help. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom: warble on registration?
Ken D'Ambrosio wrote: Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Are you talking about the warble that the phone makes every sip registration if there are messages waiting??? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
Doug Lytle wrote: »Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ You're missing your pipes, also using a comma after a 9 will give a simulated second dial tone. digitmap=9[2-9]xx|[2-9]xx|9,1[2-9]xx|[2-9]xx Actually, the pipes in the example above are correct. I want to be able to dial: 9nxxnxx and 91nxxnxx The running count can be disabled by looking in the sip.cfg for: feature.8.enabled=1 Change it to a 0. Thanks. This worked like a charm! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Questions
Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan from phone1.cfg: dialplan dialplan.1.impossibleMatchHandling=0 dialplan.1.removeEndOfDial=1 dialplan.2.impossibleMatchHandling=0 dialplan.2.removeEndOfDial=1 dialplan.3.impossibleMatchHandling=0 dialplan.3.removeEndOfDial=1 dialplan.4.impossibleMatchHandling=0 dialplan.4.removeEndOfDial=1 dialplan.5.impossibleMatchHandling=0 dialplan.5.removeEndOfDial=1 dialplan.6.impossibleMatchHandling=0 dialplan.6.removeEndOfDial=1 digitmap dialplan.1.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.1.digitmap.timeOut=3 dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan.3.digitmap.timeOut= dialplan.4.digitmap= dialplan.4.digitmap.timeOut= dialplan.5.digitmap= dialplan.5.digitmap.timeOut= dialplan.6.digitmap= dialplan.6.digitmap.timeOut=/ routing server dialplan.1.routing.server.1.address=10.0.17.8 dialplan.1.routing.server.1.port=5060 dialplan.2.routing.server.1.address= dialplan.2.routing.server.1.port= dialplan.3.routing.server.1.address= dialplan.3.routing.server.1.port= dialplan.4.routing.server.1.address= dialplan.4.routing.server.1.port= dialplan.5.routing.server.1.address= dialplan.5.routing.server.1.port= dialplan.6.routing.server.1.address= dialplan.6.routing.server.1.port=/ emergency dialplan.1.routing.emergency.1.value= dialplan.1.routing.emergency.1.server.1= dialplan.2.routing.emergency.1.value= dialplan.2.routing.emergency.1.server.1= dialplan.3.routing.emergency.1.value= dialplan.3.routing.emergency.1.server.1= dialplan.4.routing.emergency.1.value= dialplan.4.routing.emergency.1.server.1= dialplan.5.routing.emergency.1.value= dialplan.5.routing.emergency.1.server.1= dialplan.6.routing.emergency.1.value= dialplan.6.routing.emergency.1.server.1=/ /routing /dialplan Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing a group of phones but not if they are busy
Carlos Chavez wrote: I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? chanIsAvail Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page() Function Timeout
Ken Williams wrote:
I'm trying to use a simple page function. It starts a MeetMe
conference with the devices I've listed, but the devices hang up after
3-5 seconds. After doing some research I found this was a problem,
and I needed to remove a (5) from app_page.c
Well, my app_page.c didn't have the (5). I did make clean; make
install again just in case I had some weird compiled version installed
that had the (5) in it. After compiling I restarted the asterisk
service and tried paging again and still had the same problem.
In the CLI I get the following, which you can see the (5) is still in
there somehow.
-- Playing 'beep' (language 'en')
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038
-- Created MeetMe conference 1023 for conference '1010553064d'
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/717-09a48758
I've grep'd the entire src folder for \(5\) as well as qxd trying to
find all instances of this, and the only ones are listed in the
app_page.c file. Any suggestions on where to get this rogue (5) out
of here?
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
and
if (!res) {
snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd,
confid, $
pbx_exec(chan, app, meetmeopts, 1);
}
are the only sections of the app_page.c that have the meetme call in it.
My page functions, fwiw, both have the same problem:
;Paging
exten = 760,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 760,2,Page(SIP/717SIP/710SIP/702|d)
exten = 760,3,Hangup
exten = 761,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 761,2,Page(SIP/717SIP/710SIP/702)
exten = 761,3,Hangup
Any suggestions would be very helpful.
I had the same problem and ended up changing the 5 to a 300. If you
don't specify a (N) after the 'w', I believe it defaults to 5.
Steve
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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS
Zeeshan Zakaria wrote: Everything was working perfect until I updated CentOS using yum update. errors are like these. Please help, what is the solution for this. Obviously the zaptel hardware is not loading, byt why? How to load it again? Sep 13 18:33:34 VERBOSE[1490] logger.c: [chan_zap.so]Sep 13 18:33:34 VERBOSE[1490] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) Sep 13 18:33:34 VERBOSE[1490] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 13 18:33:34 VERBOSE[1490] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c: == Parsing '/etc/asterisk/zapata- auto.conf': Found Sep 13 18:33:34 VERBOSE[1490] logger.c: == Parsing '/etc/asterisk/zapata_additional.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c: == Parsing '/etc/asterisk/zapata_additional.conf': Found Sep 13 18:33:34 WARNING[1490] chan_zap.c: Unable to specify channel 1: No such device or address Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to register channel '1-23' Sep 13 18:33:34 WARNING[1490] loader.c: chan_zap.so: load_module failed, returning -1 Sep 13 18:33:34 WARNING[1490] loader.c: Loading module chan_zap.so failed! Did you rebuild the zaptel kernel drivers after upgrading the kernel??? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding custom fields (more than one) to CDR DB
Mike wrote: Hi all, I just found out how to set the column userfield, in the CDR DB to whatever I desired. Can I add multiple custom columns to the DB and fill them from the dialplan, or is it limited to one column? I am using Asterisk 1.2.4 and MYSQL for the CDR DB. As far as I know, it is just userfield. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Elpidio Ramos wrote: Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems? I am using Fedora Core 3. I have followed the instructions in several tutorials and tried several soft phones and the SIP interface seem to be dead. 1. When loading asterisk SIP load with no problem 2. When I activate the DEBUG for a peer, ip or sip in general, I don't get to see any messages when a connection is attempted from any soft phone. 3. The soft phones all report a timeout when trying to register. 4. Tried also to move to port 80 that I know is open but still the same problem. I will appreciate any help anyone can provide with this problem. Do you have iptables turned on and/or port 5060 UDP blocked? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Elpidio Ramos wrote: I am not up to speed on networking in linnux/fedora core 3. How can I verify if I have iptables turned on or the port 5060 is blocked? /etc/init.d/iptables status if you see a bunch of ports, make sure that one of them lists 5060 as ACCEPT, and that the RTP ports (defined in /etc/asterisk/rtp.conf) are also marked as 'ALLOWED'. To see if this is affecting you, you could also do a /etc/init.d/iptables stop (which stops it from running on this boot), and see if the phone works. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Elpidio Ramos wrote: Bob, I get the same answer you get when using netstat -an When I query the firewall rules I get this: Chain RH-Firewall-1-INPUT (2 references) target prot opt source destination ACCEPT all -- anywhere anywhere ACCEPT icmp -- anywhere anywhereicmp any ACCEPT ipv6-crypt-- anywhere anywhere ACCEPT ipv6-auth-- anywhere anywhere ACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353 ACCEPT udp -- anywhere anywhereudp dpt:ipp ACCEPT all -- anywhere anywherestate RELATED,ESTABLISHED ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:ssh ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:http REJECT all -- anywhere anywhere reject-with icmp-host-prohibited I assume this indicates port 5060 is restricted? Yep. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Bob Chiodini wrote: I think all anywhere should allow 5060. Try running service iptables stop (as root) to shutdown the firewall. See if 5060 then answers. I'm not running a firewall on my asterisk box so I'm not sure what the rule would need to be. service iptables start will restore the firewall. Bob... Elpidio Ramos wrote: Bob, I get the same answer you get when using netstat -an When I query the firewall rules I get this: Chain RH-Firewall-1-INPUT (2 references) target prot opt source destination ACCEPT all -- anywhere anywhere ACCEPT icmp -- anywhere anywhereicmp any ACCEPT ipv6-crypt-- anywhere anywhere ACCEPT ipv6-auth-- anywhere anywhere ACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353 ACCEPT udp -- anywhere anywhereudp dpt:ipp ACCEPT all -- anywhere anywherestate RELATED,ESTABLISHED ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:ssh ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:http REJECT all -- anywhere anywhere reject-with icmp-host-prohibited I assume this indicates port 5060 is restricted? It ought to. The example above is 'REJECT' all -- anywhere. Change the REJECT to ACCEPT and restart, and everything should be golden (for testing). If this box has any slight chance of being hacked into over the net, though, I would look at the iptables docs and lock it down. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE207P
I just bought a Te207P, and I was wondering if there is anything special that I have to do in Asterisk's zapata.conf or the zaptel.conf to enable the echo canceller, or if it is automatically enabled. Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Peers Disappearing
Douglas Garstang wrote: Can someone tell me what this is about? Asterisk seems to be 'losing' peers. Usually when a peer isn't known (such as when you first start Asterisk), Asterisk will do a database lookup and find the peer, and then seed them. I tried to dial 3254101, and I get the error below. I ran an ngrep and Asterisk isn't even doing a database query to find the peer. Why would it be doing this? It's almost as if Asterisk is expiring the phone before the phone re-registers. The phone is registering with a 900s expirey period. Asterisk has maxexpirey=3600 and defaultexpiry=900 in sip.conf. When the phone re-registers, Asterisk repopulates the peer with it's IP address. *CLI -- Executing Dial(SIP/3254101-6a9f, SIP/3254103|20|tr) in new stack Aug 18 11:59:05 NOTICE[29503]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/3254101-6a9f' status is 'CHANUNAVAIL' *CLI *CLI *CLI sip show peer 3254101 Do you have rtcachefriends=yes and rtupdate=yes set in your sip.conf? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE207P
Jeremy McNamara wrote: Steven Ringwald wrote: I just bought a Te207P, and I was wondering if there is anything special that I have to do in Asterisk's zapata.conf or the zaptel.conf to enable the echo canceller, or if it is automatically enabled. Make sure echocancel=yes is in zapata.conf. Also, you can look in dmesg and see if the VPM came online with the kernel module load. Ok. Here is what dmesg has to say: TE2XXP: Span 1 configured for ESF/B8ZS wct2xxp: Setting yellow alarm on span 1 SPAN 1: Primary Sync Source VPM400: Not Present OCT Result: 1234/5678 Before chip open! After chip open! wct2xxp: Clearing yellow alarm on span 1 VPM450: Present and operational servicing 4 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for ESF/B8ZS wct2xxp: Setting yellow alarm on span 2 Completed startup! I am assuming that means it is operational. In zapata.conf, the PRI that is connected has this: echocancel=yes echocancelwhenbridged=yes echotraining=yes Still hearing echo on the line, though. Is there anything additional that I need to do? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys and Call Park
Has there been any progress on getting Call Parking to work with Linksys SPA-942 phones and Asterisk? I am willing to assist, if there are people working on this already. I have done a little research on this, and it looks like there are people asking for it, just haven't found anyone *doing* it, and don't want to wind up duplicating effort. Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More SNOM, Message Indicator/Retrieval issues
J. Oquendo wrote: I've got a strange issue with SNOM's and Asterisk v.1.2.10 [EMAIL PROTECTED] ~]# asterisk -rx show version Asterisk 1.2.10 built by root @ comp on a i686 running Linux on 2006-07-24 23:42:12 UTC Verbosity is at least 10 Core debug is at least 1 My SNOM's are a mixture of 360's and 320's. Before hand here is my entry for voicemail: exten = default,1,VoicemailMain() exten = asterisk,1,VoicemailMain() exten = unknown,1,VoicemailMain() exten = Unknown,1,VoicemailMain() Now this is what is happening. I leave a message, less than a minute later, Message light on the SNOM lights up. Hit the retrieve button, asks me for my username and password. Enter 3200 for username, 3200 for password, it tells me it's an incorrect login. I hang up. Hit the retrieve button, user 3200 password 3200. You have X_Amount of messages. It is giving me bad information on the initial retrieval but allowing it in on the second attempt. Anyone else experience this? Also, has anyone been able to successfully get the sidecar working with these phones. I have one 360 with a sidecar and the only buttons that illuminate are for those phones which are unregistered. It should be showing me who is on the phone. What do you have dkey_retrieve set to on the phone? Mine is set to speed sip:[EMAIL PROTECTED];user=phone (extension/IP of the asterisk box). Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-942 TFTP Provisioning
Jeremiah Millay wrote: I'm trying to provision some spa-942 phones via TFTP. The phones get their address from a dhcp server which sends it option 66 (address of the tftp server). After spending some time with the phones and even breaking down to sniff traffic from the phones I see that they are not requesting their config from tftp. I can kind of fake the phones into grabbing their configs by doing something like: http://192.168.20.77/admin/resync?tftp://X.X.X.X/spa000e08db9208.cfg This will provision the phones correctly but it requires my intervention. My configs are based on those taken from this site: http://voipspeak.net/index.php?option=com_contenttask=viewid=73Itemid=28 I'm confident that on the tftp server side everything is correct since it works when I force the resync. In my network environment I have a number of Cisco IP phones that get their configs from the same TFTP server and receive the same info from the DHCP server and they are working correctly. So tftp is running and dhcp is leasing out the correct info. Anyone run into this problem with the spa-942 or similar model? Any help would be greatly appreciated. One thing that messed me up with them is having stray '' or '' characters in the file. CallerID is an example. This *really* confused the XML parser on the phone. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting CALLERID on a residential telco line
hugolivude wrote: Redhat 9 Asterisk - 1.2.7 TDM 400 - 1 FXO, 2 FXS I'm using a standard residential PSTN line on my ZAP channel and curious whether I can override the caller ID my telco has for me with one of my choosing. I've tried this: exten = s-ZAP,n,Set(CALLERID(all)=My Name 999-999-999) exten = s-ZAP,n,Dial(Zap/g2/6137451576) but the callee still sees my telco callerid. Have I missed something or does the telco ultimately control CallerID on a residential line? It stands to reason it would, but I'm hopeful I'm wrong!! If it is a POTS line, you cannot change the caller*id. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SipAddHeaders Question
I have added a header using the SipAddHeader command. At some point later, I would like to clear this header, as I no longer need it. For instance, I add the Call-Info header for auto-answering a SNOM phone. When I transfer the call to another snom phone, the auto-answer header travels along with the call, resulting in another auto-answer. What I would like to have happen is the header be cleared. I did a quick google search, and didn't find anything about this. Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mail loop?
Mike Fedyk wrote: Is anyone else getting messages from the lists.digium.com mail server with errors about a mail loop? I've been getting this for the last few weeks, but I don't have any list software on my server. Any ideas? Yep. I have been getting them quite steadily today. Looks like every email I ever sent to the list is coming back to me now. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Include Text file in Dial Plan
Forrest Beck wrote: Is there a way to include a search of a text file in the dial plan? I am trying to think of a good way to keep a sort of Blacklist file that is checked against before letting a call through. If the callerid is listed in the file, it will go to Hangup() From http://www.voip-info.org/wiki-Asterisk+config+extensions.conf One big file or several small? With the *#include filename* statement in extensions.conf, other files are included. This way you can setup a system where extensions.conf is the main file, *users.conf* contain your local users, *services.conf* contain various services, like conferencing. This way, the dial plan may be easier to maintain, depending on the size of your setup. The *#include filename* statement is *not* the same as the *include context* statement. The *#include* statement works in all Asterisk configuration files http://www.voip-info.org/wiki/index.php?page=asterisk+config+files. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to hang the zap channel
soft hangup Zap/4-1 Steve Bartosz Wegrzyn - asterisk wrote: After all users disconnect the Zap channel is still connected to pSTN call. asterisk1*CLI show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Up MeetMe(500|xApMs|1234) Zap/pseudo-141305407 [EMAIL PROTECTED]:1Rsrvd (None) 2 active channels 1 active call I wish I could shutdown that channel when all users disconnect. Maybe I am doing something wrong. Maybe the extensions design is wrong. Thanks On Thu, Jun 15, 2006 at 10:37:23AM -0500, Bartosz Wegrzyn - asterisk wrote: in which extension, the thing is that when every (voip) user disconnects , the zap channel is still connected to the conference, How about a nice little show channels ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comedian Mail not deleting .txt file
Matt wrote: I too have seen this happen on two occassions. Said there was a 93 second message (when I logged into the web interface).. commedian mail said there was a message, but there was nothing. 1.2.7 See the following: http://bugs.digium.com/view.php?id=7125 Fixed in 1.2.9. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling zaptel on FC5
J.J. Feminella wrote: Are there any generic install guidelines for compiling the Zaptel drivers on FC5? This is my first install of Asterisk (and my first FC5 system) and I'm having a great deal of trouble getting it to cooperate. make clean and make are definitely not playing nice, telling me that You don't appear to have the kernel sources installed when I'm pretty sure that I do. Any pointers? Make sure you have the kernel-devel package installed for the currently running kernel. make ; make install in the libpri directory (if you need PRI support) make linux26; make install in the zaptel directory make ; make install in the asterisk directory This is the basic procedure that I follow when I install/upgrade to a newer version of Zaptel/libpri/Asterisk. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Vs SIP cpu load
Mike Lynchfield wrote: taskset does not seem to exist on redhad 9 nor freebsd.. ;) On Fedora Core 4, it is provided by the schedutils RPM. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom high SIP ping time
Steve Glaus wrote: Mike Hammett wrote: I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify was returning ping times anywhere from 20 to 70 ms over a sparsely used LAN. Command prompt (ICMP) pings were under 1 ms. No amount of different Asterisk versions or phone firmware revisions seems to solve this. All was well, then (as far as we know) without changes, it crapped out. I'm having much the same issues only I'm using Cisco 7960 phones. When I do a 'sip show peers' I'm getting times in excess of 300ms. A soft phone on the same network (x-lite), is reporting times of 4 ms. Related to this (I think), I'm getting audio issues. The person being called can hear the caller fine but the callee's voice drops in and out excessively. I have qualify set to yes in the sip definitions for all the clients (Including the soft phone). Does anyone know what is causing this. I'm not aware what the sip ping times were earlier, but the audio issues seemed to have started spontaneously. Anyone have any idea regarding this? What codecs are you using? I have noticed that g729, for some reason, adds a lot of latency to the phone. Running on uLaw, however, I get times from sip show peers of around 5-14ms. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config Revision Control
Bruce Reeves wrote: I setup a subversion server and a trunk for my different server configs. You might look at that, it does not appear to keep file level versions, but it works great here. On 6/2/06, *Douglas Garstang* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. I do something like this with subversion, except that I have a set of common files that hardly ever change, and then files that are specific to the machine. The ones that are specific to the machine I use the 'include' functionality to put into the main files. Something like this *might* help you out. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, voicemail, no codec_g729
Kristian Kielhofner wrote: I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only format=g729 in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the message is supposed to be recorded, the voicemail app bombs and this is displayed on the console: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! From what I understand, that is the format that Asterisk uses internally. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrading
Chris Blunt wrote: Hi List, I was wondering what is the best way to upgrade an Asterisk system to the latest version. I know there is the patch method, but if I am jumping 3 or 4 versions is a re-install the best way? Should I just make the files then manually copy them in? Does this avoid overwriting any modified sound files etc? Should I delete the current files or move / make a copy to a different location first? I know this is a lot of questions but I am hoping for a best practice idea etc… I believe that make upgrade installs just the applications, and does not touch config files (which are only installed with make setup, BTW) and the sound files. Hope this helps. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting the Server IP
Hello all! Can anyone think of an *easy* way to get the IP number of the server running asterisk from within the dialplan? Thank you in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 w/ 1.1.0.11 firmware
I had provisioning via tftp working on this phone. I have verified that after the firmware upgrade, it contacts the tftp server and downloads the cfgMACADDR file, and the ring/etc files successfully. Unfortunately, changes made to the config file don't make it to the phone (SIP account info/server info, etc). The script that I am using to generate the binary files is loosely based on this script: http://www.voip-info.org/users/557/15557/images/433/config.pl.txt I also have a phone running the stock 1.0.1.12, and it comes up just fine, so I am pretty sure that it isn't the script. Does anyone know if Digium changed the format of the config? Is anyone else out there running 1.1.0.11 with tftp provisioning working? Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM autoanswer question
I have run into a small snag with SNOM phones, asterisk, and autoanswer. I direct an extension to autoanswer a SNOM 320 phone. Call is autoanswered, and call progresses correctly. I then execute the agi command to transfer to another SNOM 320. Unfortunately, Asterisk does not clear the Autoanswer Call-Info string, and the second phone will also autoanswer. Is there anyway that I can reset the Call-Info: field in the SIP header? Thank you in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User Defined VoiceMail announcement?
I think that he meant recording the unavailable and busy messages for the mailbox. To do this, log into the voicemail-box, and hit '0' for Mailbox options. The options that you are interested in are numbered 1-3. Steve Dovid Bender wrote: Well said. Or you can create an extension to which people dial in to to check thier VM Exten 8000,1,Voicemailman() --- C F [EMAIL PROTECTED] wrote: RTFM On 4/24/06, Benoit Panizzon [EMAIL PROTECTED] wrote: Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Grüssen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
Steve Kennedy wrote: On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote: D-link has a nice one, optional 5 year warranty on some of the commercial stuff Though beware, some of the D-Link ones only have half the ports with PoE. Actually, as far as I know, only one of the D-Link POE switches is like that, the DES-1316 has 16 ethernet ports, with 8 of them POE. The 1516 has 26 POE ports (with two of them gig). Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
William M Conlon wrote: Beware. Although the Polycom 501s will sink power over ethernet (that is they are powered by a cable pair within a cable that resembles ethernet), they are NOT IEEE 802.3af POE devices! They work on 12 volts (I think -- haven't measured it) instead of 48VDC. So don't expect to buy a POE source and expect the phone to receive power just by plugging in a patch cable. They must be powered by the Polycom voltage sources. Nevertheless, here's what works for me: Netgear FS108 :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 Some notes: 1. The Polycom injector cable should be plugged into a POE port on the switch (the Netgear FS108 switch has both powered and unpowered ports), or the Polycom injector will not source power. 2. The Netgear FS108 is NOT sourcing power. 3. The patch cable is a 50-foot CAT5. 3. To beat a dead horse, the Polycom 501 itself, is NOT a POE phone, IMHO. Caveat emptor. They are 802.3af, if you buy the correct cable (they two different types): (From http://www.voip-info.org/wiki-Polycom+Phones) (phones with the cables) SoundPoint IP 501 (NA PSU)|2200-11531-001|$270 SoundPoint IP 501 (IEEE PoE)|2200-11531-025|$295 (just the cables) NA PSU for 30x,50x,600 Qty 5|2200-07496-001|$35 IEEE PoE cable for 30x,50x|2200-11077-002|$35 Cisco PoE cable for 30x,50x|2200-11014-002|$35 I believe the reason the 30x and 50x's are like this is because the standard was still in flux when they were designed. They put the POE brains into the cable to make it easy to switch to whatever standard was decided upon. The 601 comes with built in POE (compatible with Cisco and IEEE). Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly
Bryan Boatright wrote: I too am experiencing DTMF problems with 1.2.7.1 that I did not experience with recent prior versions. I've backed up to version 1.2.6 and so far DTMF detection is working reliably (but that's only with about 10 calls worth of testing). I've only had problems over SIP channels. Zap channels did not have problems with 1.2.7.1. I do not have any IAX channels, so cannot comment on that. I know others tend to discount DTMF problems because of known problems with how Asterisk handles DTMF, but there does seem to be enough anecdotal evidence that something bad has recently happened to make things worse. Dave, would you mind trying version 1.2.6 to see if that also resolves your problems? I hate to say me too, but I have been experiencing some DTMF issues since 1.2.4. Have tried with 1.2.4, 1.2.6, and 1.2.7.1; all with the same result. No DTMF, regardless of SIP INFO, RFC2833, or inband(ulaw). This is on a inbound SIP trunks from Level3. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enablling Te110p with PRI
Rafael Visser wrote: Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) Question: Is there a how to connect the Asterisk to an ericsson sw? What other test can i do against the switch?. Thanks in advance... What does the exact Dial line look like in your extensions.conf? Is 0971200152 the number that the other end is expecting? For instance, our Shoretel requires the country code be added, for instance 1503XXX. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register question
I am trying to link an asterisk box up to a SIP server on the same subnet. The SIP server does not have a password (and is locked down by IP number 'allow'). How do I specify this on the register line? Based on the documentation, the line looks like this: register = user[:secret[:[EMAIL PROTECTED]:port][/extension] It looks like [EMAIL PROTECTED] is the minimum required. Is there anyway to specify a username of null, or something? Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [1.2.5] DTMF not being set correctly (RESEND)
Rich Adamson wrote: I am having trouble getting DTMF mode to be set to inband on incoming calls. I have the following set, and for some reason the connection is still negotiated with rfc2833. [outbound] type=friend secret=XXX username=XXX authuser=XXX host=XXX.XXX.XXX.XXX context=inbound qualify=200 insecure=very disallow=all allow=ulaw dtmfmode=inband dtmf=inband canreinvite=no nat=no Since you didn't mention what device/itsp is generating the incoming call, I'll have to assume you are trying to use sip with an external itsp. If that's a correct assumption, the itsp may not support inband, suggesting the session is rfc2833 only. The ITSP, Level3, claims that they support it (for U-Law), and that my end is not negotiating it (as indicated by a 101 in the SDP). I can see the 101 when I do a SIP debug, but no matter what I do, the 101 does not go away. You might want to try 'sip debug' and see what's happening, or 'set verbose xx' where xx is some large number like 30 or so. One or the other should provide more detail. Right. That is how I have verified that my end is not setting the dtmfmode correctly. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [1.2.5] DTMF not being set correctly (RESEND)
I apologize if this gets posted twice. Tried once about 5 or so hours ago, and still have not seen the message on the list I am having trouble getting DTMF mode to be set to inband on incoming calls. I have the following set, and for some reason the connection is still negotiated with rfc2833. [outbound] type=friend secret=XXX username=XXX authuser=XXX host=XXX.XXX.XXX.XXX context=inbound qualify=200 insecure=very disallow=all allow=ulaw dtmfmode=inband dtmf=inband canreinvite=no nat=no Is there something that you can recommend that I add to get this to work? I am running 1.2.5, and the rfc2833 mode works like a champ, but I would like to be able to support inband, too. Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH native files
Tomislav Parčina wrote: Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! You need to use mpg123 to convert the mp3 files to wav files first. mpg123 -w out.wav in.mp3 sox out.wav -r 8000 out.gsm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ImportVar Syntax
I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's Direct Bridge variable. I have tried: ImportVar(TEST=SIP/25-6d2a|name) which doesn't seem to do anything. Looking through the code, the thing that I am looking for is: c-_bridge-name (in function handle_showchan). The voip-info page for ImportVar returns an error, and I couldn't find any occurance of ImportVar, except in pbx.c. Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many TDM2400P's will a server take?
Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will be able to service many more interrupts/card/channels than a 500 mHz Celeron. :-) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 320 echo problems
Nora Lavelle wrote: Hi there – I’m having some echo problems on my snom 320 phones. Anybody experience this before ? I don’t have any issues with the sipura 841s I have though. Any help is greatly appreciated. Thanks ! What version of FW is the Snom 320 running? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 320 echo problems
On Thu, 2006-01-26 at 11:50 -0800, Nora Lavelle wrote: Thanks so much for your help. The version information is attached below. Looks like it is 4.4 Also can you tell me which exact Setting under Audio in Advanced Options I should change ? Here are the options I see. [snip] Firmware:http://snom.com/download/share/snom320-4.4-SIP-j.bin Production Information:Mac:0004132425F6;Version:Standard;Hardware:snom320 (MB V1.0_K7,KB V1.0_L4-NC);Lot: 11/05 Not sure about the gain values, but to update the firmware, go to the web interface and use the following URL: http://snom.com/download/snom320-5.2a-SIP-j.bin The 'a' version of the firmware is for non 5.X version upgrades. Hope this helps some! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI restarting each hour?
Michael Collins wrote: Has anyone else had this symptom? It seems like my PRI “restarts” each hour. I’ve got a PRI from Qwest in central California. What I don’t know is if this is on the * side or on the telco side. I don’t use my system enough to know if this is a big problem, a minor glitch or just an “undocumented feature” - ☺ Any suggestions on how to track the source of these “restarts?” Anybody else seen something like this? It is normal/default behaviour for Asterisk/Zaptel to unused PRI channels on an hourly basis. The following, from voip-info.org, describes the zapata.conf entry that will change this behaviour. *resetinterval*: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. Hope this helps. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 601 Bricked?
I have a Polycom 601, and it seems to be totally bricked. When I power it on, all the light come on and stay on. The LCD never lights up. Is there any way to recover from this? Thank you in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata.conf and Realtime
I asked this a few days ago, and haven't gotten an answer (or seen my message in the archive, yet). Since there were some email problems the other day, I will just pose the question again. I would like to know if there is a way to have a table, like zapata_conf in a DB, and have asterisk realtime pull the information out, like it does for voicemail, sip.conf, and iax.conf, etc. If anyone has done this and has a schema that I could use, I would be very happy. TYVMIA Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loading zaptel drivers automatically upon reboot
On Fri, 2006-01-13 at 19:50 -0500, hugolivude wrote: Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load automatically? I've looked everywhere! - I tried running ztcfg but it did nothing, - I read a posting that spoke of editing rc.modules file, but I don't seem to have that file, - I tried removing everything that corresponds to zaptel, (including, but not limited to 'ztcfg', 'tor2' and 'tormenta' devices) from /etc/modules.conf. Again no luck Any ideas? For my TDM400 cards, I need to run the following commands: modprobe wcfxo modprobe wcfxs udevstart sleep 5 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata.conf and Realtime
Sorry if this is a double-posting. I tried sending the following message Friday afternoon, but it still hasn't made it to the list. Based on the comments in the extconfig.conf file, zapata.conf *should* support being loaded realtime. Has anyone succeeded in doing so, and what does the schema, etc look like? Thanks! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy inaccuracy on linux-2.6
On Mon, 2006-01-16 at 17:43 +0100, Tamas wrote: Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 136 passes --- Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975853 linux-2.6.15, zaptel/branches/1.2 r900, RTC Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 96 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952942 linux-2.6.15, zaptel/branches/1.2 r900+patch bugs.digium.com/view.php?id=5971, RTC Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 136 passes --- Best: 100.00 -- Worst: 99.694824 -- Average: 99.951973 HW: Tyan Tomcat K8E, Athlon64 3000+, 1GB RAM, 3ware 8006, 2x Maxtor HDD SW: Ubuntu 5.10, linux-2.6.15, zaptel from 1.2 branch Any idea what can be wrong? What does your /proc/interrupts say? On my asterisk box, I was seeing crappy interrupt handling like this only when I was using XT-PIC interrupt handling, when I moved to IO-APIC, things got much better... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime voicemail
On Sun, 2006-01-15 at 23:02 -0800, [EMAIL PROTECTED] wrote: i tried to setup realtime voicemail recently with 1.2.1 but couldn't get it to work. no matter what i do. it still looks for config in the voicemail.conf file. (BTW realtime sip extensions works fine) here's the voicemail line in extconfig.conf: voicemail = mysql,asterisk,voicemail here's the mysql schema: CREATE TABLE voicemail ( uniqueid int(11) NOT NULL auto_increment, customer_id bigint NOT NULL default '0', context varchar(50) NOT NULL default '', mailbox bigint NOT NULL default '0', password varchar(10) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, attach varchar(3) NOT NULL default 'yes', saycid varchar(3) NOT NULL default 'yes', hidefromdir varchar(3) NOT NULL default 'no', PRIMARY KEY (uniqueid), KEY mailbox_context (mailbox,context) ) TYPE=MyISAM; am i missing something? That looks like the minimal config... Something I found out was that I need to set the context = '' for it to work. I was using default, but for some reason I could never get that to work. Perhaps it is a bug? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List
On Mon, 2006-01-16 at 15:28 +1100, [EMAIL PROTECTED] wrote: The list is very quiet today - almost too quiet Yes, I have noticed the same thing. I have sent about 4 or 5 messages to the list, and the first one I sent (about 5 hrs ago) has yet to arrive. Perhaps there is something going on with the list-serv? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy inaccuracy on linux-2.6
On Mon, 2006-01-16 at 21:51 +0100, Tamas wrote: cat /proc/interrupts CPU0 0:6645053IO-APIC-edge timer 1: 8IO-APIC-edge i8042 2: 0 XT-PIC cascade 5: 3309 IO-APIC-level eth1 7: 679362 IO-APIC-level eth0 8:8338011IO-APIC-edge rtc 10:204 IO-APIC-level eth2, HFC PCI 11: 20559 IO-APIC-level 3w- NMI:404 LOC:6644437 ERR: 0 MIS: 0 eth2 is not use currently. This box is in preparation for production. I don't know how can the HFC PCI card (Billion 1xBRI) get the same IRQ as eth2 [onboard Broadcom NIC]. Probably because it's on different bus: :04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at fe5f (64-bit, non-prefetchable) [size=64K] Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [58] Message Signalled Interrupts: 64bit+ Queue=0/3 Enable- Capabilities: [d0] #10 [0001] :01:08.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 10 I/O ports at d000 [disabled] [size=8] Memory at fdffc000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 Anything else to take a look for? Ok. That looks like it *should* be working correctly. It is interesting that your cascade interrupt is still XT-PIC, and the highest interrupt listed is 11. I have attached the output from my /proc/interrupts for comparison. Does the bios have any mention of APIC/legacy or anything??? The board I am using is an Asus K8S-mx with a Sempron64 2800+ in it... Steve CPU0 0: 488733IO-APIC-edge timer 1:266IO-APIC-edge i8042 4: 4175IO-APIC-edge serial 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 169:1853920 IO-APIC-level libata, wct2xxp 177:492 IO-APIC-level eth0 185: 0 IO-APIC-level SiS SI7012 193: 9869 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217: 0 IO-APIC-level ohci_hcd:usb4 225:1850310 IO-APIC-level wcte11xp NMI:144 LOC: 488707 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy inaccuracy on linux-2.6
On Mon, 2006-01-16 at 22:30 +0100, Tamas wrote: Hello Steven! Thanks for answers and suggestions! I will check the bios again tomorrow. I found some interesting things in /var/log/dmesg: ... CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 512K (64 bytes/line) mtrr: v2.0 (20020519) CPU: AMD Athlon(tm) 64 Processor 3000+ stepping 02 Using IO-APIC 2 ..MP-BIOS bug: 8254 timer not connected to IO-APIC works. Hm. This looks like something here (above) pcie_portdrv_probe-Dev[005d:10de] has invalid IRQ. Check vendor BIOS assign_interrupt_mode Found MSI capability Allocate Port Service[pcie00] Maybe this says somehting... Yeah. Also, what does your kernel config look like? Why did you roll your own kernel, rather than using the kernel 2.6.15 series that is in the updates-testing??? The AMD64 box that I am using is running 2.6.14-1.1656_FC4, btw. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Decent sub-$100 SIP phone.
On Mon, 2006-01-09 at 15:28 -0500, Ken D'Ambrosio wrote: Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like Princess phones, and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy environment (a cleanroom), the less overall static, the better. Someone suggested the Polycom 301's, but I'd lose money on them. [I'll go with them if I have to, as I'm making money elswhere, but still...] So, does anyone have any suggestions for decent sub-$100, professional-looking SIP phones? If you were looking at BudgeTones, you *might* want to look at the GXP-2000. A little nicer, and if you shop around you can get them for a decent price. http://snipurl.com/lfa3 for instance. The Polycom is nice, but I have found that the only Polycom that seems to do PoE correctly is the 601, which is definitely out of your sub-$100 price-range... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 360 locked up
Hello all!
I was trying to get the dial-string setup for my regular usage, and the
phone locked up in the middle of dialing. Basically, I put the following
line in, hit save, and got as far as dialing '9', and the phone froze.
|^(9[0-9]{10}|sip:[EMAIL PROTECTED]|d
Now the phone boots up to the SNOM splash screen and hangs there. I can
ping it, but cannot get to the web-interface and cannot reset to factory
defaults using the web-gui.
Any idea how I can reset the phone to factory w/o using the GUI? Or am I
completely hosed?
Steve
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RE: [Asterisk-Users] SNOM 360 locked up
On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote: Try loading http://phone-ip-address/line_sip.htm?settings=saveuser_dp_str1= (if that was in the line 1) while the phone boots up (keep your finger on the reload button). If that does not work, you need to do a tftp update. Yeah. The website address didn't work. (The phone, I think, is not far enough along to even start the webserver). I will try the tftp update method, and see what happens. So far, though, it doesn't seem to be hitting the tftp server that I set up manually. Also consider moving to version 4.5 (http://www.snom.com/snom360_release_notes.html). Any idea how to do that? I think it is running 4.1. I have put the firmware image URL into the upgrade line before, and it didn't take. (Ended up going back to what it had previously had). Thanks for the help! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 360 locked up SOLVED
Thank you so much for your help, Christian! Your suggestion worked perfectly, and the phones came back up without a problem. Froehe Weihnachten! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2-rc1 and sip show inuse
I apologize if this question has been asked before. Did something change with the behaviour of the 'sip show inuse' command between 1.0.9 and 1.2-rc1? I used to be able to see a list of extensions and the number of in/out calls. Now it just reports: asterisk*CLI sip show inuse * User name In use Limit * Peer name In use Limit no matter how many calls are being used. asterisk*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.70.128 1234339ad96826e 00102/0 ulaw No Tx: ACK 192.168.70.116 1235723e1612-52 00101/2 ulaw No Rx: ACK 2 active SIP channels Any info about getting the previous functionality back would be greatly appreciated. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GXP2000 tftp config
Hi! I am trying to configure a series of Grandstream GXP2000 phones. I have downloaded the Grandstream Configuration Generator v1.3 to generate the cfgmacaddr files that the phone expects to see on the tftp server. I watch my tftp server's diagnostic output, and verify that it is downloading the config files. When I check the phone's web configuration, however,I find that nothing has changed on the phone. The phone reports on the status page: Program-- 1.0.1.12Bootloader-- 1.0.1.2. The config file that I am running with the program is attached. Any help getting this to work is appreciated! Steve P270 = Grandstream1 P29 = 0 P2 = password P30 = time.apple.com P31 = 1 P33=*98 P34=asterisk P35=1142 P36=1142 P3: Grandstream1 P41=192 P42=168 P43=70 P44=10 P47=192.168.70.10 P50=1 P52 = 1 P63=1 P64=240 P65=0 P72=1 P73 = 2 P74=1 P75 = 1 P78=1 P8=1 P81 = 1 P99=1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100 = T100 Upgrade
I am looking to upgrade my asterisk server from using a single analog X100P card to a T100P card. The PRI is already in the process of being ordered, and I am wondering if there are any gotchas that I should be aware of. Also, is there any reason, other than the number of ports per PCI card, for getting the TE4xxP over the T100P? Thanks in advance, 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] Voicemail/Playback Questions
We are using a SCSI based IBM eServer x300 for our PBX. In setting this unit up, we used a backup machine, which was IDE only. The problem that we are currently experiencing is that the voicemail prompts are coming out the system so fast that the words overlap each other, and sometimes are unintelligable. For instance: The person at extension 7-0-0-1 is unavailable might come out as The at 7-0-1 unavailable. This issue appears unique to the SCSI system, and did not occur with the IDE-only machine. It also is not limited to just voicemail, but all files run through Playback() /proc/cpuinfo on the SCSI machine indicates: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Celeron (Coppermine) stepping: 10 cpu MHz : 951.714 cache size : 128 KB ... and on the IDE machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 3 cpu MHz : 867.418 cache size : 256 KB The config files are the same across the two systems, and both are running the same version. Show version in the asterisk console reads: Asterisk CVS-01/30/04-19:07:39 Any help that you can provide would be appreciated. If there is any further information I am forgetting, please let me know. Thanks in Advance! Steve Ringwald ___ 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] Voicemail/Playback Questions
Tilghman Lesher wrote: On Sunday 08 February 2004 06:27, Steven Ringwald wrote: We are using a SCSI based IBM eServer x300 for our PBX. In setting this unit up, we used a backup machine, which was IDE only. The problem that we are currently experiencing is that the voicemail prompts are coming out the system so fast that the words overlap each other, and sometimes are unintelligable. For instance: "The person at extension 7-0-0-1 is unavailable" might come out as "The at 7-0-1 unavailable". This issue appears unique to the SCSI system, and did not occur with the IDE-only machine. It also is not limited to just voicemail, but all files run through Playback() Sounds like one of your libraries is buffering output and is returning too soon. Are you running exactly the same distribution/version on each? Perhaps one got an online update and the other did not? That's the only thing I can think of that would cause this type of trouble. I wouldn't suspect hardware differences, as it sounds like you're using Digium hardware for both, where it matters. Yes. Fedora Core 1 on both systems. Same version of Asterisk on both machines. (I copied the source directories of one to create the other). I have also tried updating both to the same version of Asterisk 0.7.2 (CVS), with the same results. Yes, Digium hardware (X100) is in both systems. (Actually, the same card was in both systems). The card is on its own interrupt, also: [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 0: 16557048 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci, usb-uhci 7: 165254978 XT-PIC wcfxo 8: 1 XT-PIC rtc 10: 4612230 XT-PIC eth0 11: 245282 XT-PIC aic7xxx 15: 1 XT-PIC ide1 NMI: 0 ERR: 0
Re: [Asterisk-Users] OT Superbowl = Linux Shake up to the world..
Adam Goryachev wrote: [EMAIL PROTECTED] wrote: Linux, Shake up the world oops sorry, test... testing 123 [EMAIL PROTECTED] wrote: I laughed out loud, and then looked around at all the other people in the room who were staring at me because they didn't understand the significance of the statement. For those who haven't seen the advert (I assume this is about an ad played at the superbowl) could you perhaps include a little more detail... IFilm.com provides the superbowl ads in Real, Quicktime, and WMV format: http://www.ifilm.com/?sctn=collectionspg=superbowl2004 It is on the right hand side, under IBM. Steve
[Asterisk-Users] Voicemail/Playback Questions
We are using a SCSI based IBM eServer x300 for our PBX. In setting this unit up, we used a backup machine, which was IDE only. The problem that we are currently experiencing is that the voicemail prompts are coming out the system so fast that the words overlap each other, and sometimes are unintelligable. For instance: The person at extension 7-0-0-1 is unavailable might come out as The at 7-0-1 unavailable. This issue appears unique to the SCSI system, and did not occur with the IDE-only machine. It also is not limited to just voicemail, but all files run through Playback() /proc/cpuinfo on the SCSI machine indicates: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Celeron (Coppermine) stepping: 10 cpu MHz : 951.714 cache size : 128 KB ... and on the IDE machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 3 cpu MHz : 867.418 cache size : 256 KB The config files are the same across the two systems, and both are running the same version. Show version in the asterisk console reads: Asterisk CVS-01/30/04-19:07:39 Any help that you can provide would be appreciated. If there is any further information I am forgetting, please let me know. Thanks in Advance! Steve Ringwald ___ 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] Distinctive ring Issues
Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0 dring1context=dist_ring1 dring2=95,325,95 dring2context=dist_ring2 dring3=325,0 dring3context=dist_ring3 ; If no pattern is matched here is where we go. context=dist_ring0 channel = 1 I am assuming that 95 ms is a short ring and 325 ms is a long ring. In my extensions.conf file, I have the following contexts defined: [dist_ring1] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7002,SIP/ringwald) [dist_ring2] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7003,SIP/ringwald) [dist_ring3] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7005,SIP/ringwald) [default] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7001,SIP/ringwald) No matter which number I dial, I always get the [default] context on answer. Can anyone shed any light on what I am doing wrong? The PSTN line is through Qwest Business, and uses US format distinctive ring tones. Show version in the asterisk console returns: Asterisk CVS-01/27/04-19:07:39 Thank you in advance for any help! Steve Ringwald ___ 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] Grandstream 101
dkwok wrote: Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060 dtmf = sip info codec = pcmu codec = pcma Any pointer of a sample of config file would be most appreciate. Here is what my sip.conf file looks like for a grandstream phone: [sringwald] disallow=all host=dynamic allow=ulaw type=friend username=sringwald secret=SOME SECRET callerid=Steve 77 canreinvite=no reinvite=no insecure=yes nat=yes dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=77 ; Mailbox for message waiting indicator ___ 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 + BudgeTone (behind NAT)
Steve wrote: On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote: I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from me. From the console "sip debug" shows that the SIP part is working fine and DTMF via SIP INFO works. I use OpenBSD firewalls with NAT and redirect and it works just as it's supposed to. That's not even half duplex. In half duplex each side Can talk, but only one at a time. It seems to be an error with configuring your firewall. (One common error is to only turn on redirect. But you also need to Allow the traffic to flow... I am having problems similar to Owen's. Just for grins, can you tell me which ports you opened up? I opened the following: tcp 4569 192.168.2.212 udp 4569 192.168.2.212 udp 5036 192.168.2.212 udp 5060 192.168.2.212 tcp 2:21000 192.168.2.212 udp 2:21000 192.168.2.212 192.168.2.212 is the IP of the Asterisk box within my firewall. I have no trouble connecting to it on the local LAN, but if I go remote, it always wants to connect via a bridge connection to the other BudgeTone phone. sip.conf [ringwald] fromuser=ringwald disallow=all host=dynamic allow=ulaw type=friend username=ringwald secret=MySecret canreinvite=no reinvite=no nat=yes dtmfmode=inband ; Choices are inband, rfc2833, or info Any help that you can provide would be greatly appreciated. Steve Ringwald
