[asterisk-users] How to set the II DIgits?
I need to set the II digits for some outgoing calls originating with asterisk, but the documentation seems to show that all the various ANI2 variables are read-only. So how do I set them? (Yes, we have Feature Group D trunks and allowed to set them and regularly do with our C.O. switch. The interface between that switch and asterisk is via ISDN spans.) Thanks... Jim Gottlieb San Diego, California ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 Change the above to host=dynamic I just did this and did a 'reload'. reg.1.server.1.address=jtsd05 Can the phone resolve this unqualified name? Yes. It's in the search path, but just to be sure I put in an FQDN. Still, no change :-( ... chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed for '192.168.200.99' - Username/auth name mismatch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On 2009-06-15 at 19:12, Jeff LaCoursiere (j...@jeff.net) wrote: chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed for '192.168.200.99' - Username/auth name mismatch I am a bit confused as to the names and addresses involved here. Which name/address is the server, and which is the phone? The phone is 192.168.200.99. The server is jtsd05. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On 2009-06-15 at 17:04, Dave Fullerton (dfullertaster...@shorelinecontainer.com) wrote: Try changing reg.1.address to hft0. My hunch is asterisk is looking at the from of 6193644...@jtsd05 and going huh? I don't know a 6193644...@jtsd05. That makes sense and it fixed it. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom registration errors
I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from 'sip:6193644...@jtsd05' failed for '192.168.200.99' - Username/auth name mismatch Turning on SIP debug, it appears it's asterisk trying to register with the phone: Using latest REGISTER request as basis request Sending to 192.168.200.99 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.200.99:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99 From: 6193644850 sip:6193644...@jtsd05;tag=A1BB38FF-7161AAEA To: sip:6193644...@jtsd05;tag=as3d68239c Call-ID: 20f907fe-db323389-f4569...@192.168.200.99 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 But then, the From: and To: lines seem to both show it from hostname jtsd05, though there's also the line saying it's going to 192.168.200.99 (the phone). I've played with all sorts of settings in sip.conf, but the messages persist. Here's what I've got: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 disallow=all allow=ulaw dtmfmode=rfc2833 progressinband=no ;Polycom phones have trouble with the progressinband=never callerid=HFT Booth 0 (619) 364-4850 allowsubscribe=yes And some of the Polycom phone config: reg reg.1.displayName=HFT0 reg.1.address=6193644850 reg.1.label=4850 reg.1.type=private reg.1.lcs= reg.1.csta= reg.1.thirdPartyName= reg.1.auth.userId=hft0 reg.1.auth.password=mysecret reg.1.auth.optimizedInFailover= reg.1.musicOnHold.uri= reg.1.server.1.address=jtsd05 reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires= reg.1.server.1.expires.overlap= reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.server.1.lcs= reg.1.outboundProxy.address= Any ideas would be welcomed. Thanks... ...Jim Gottlieb, San Diego, California ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hostname in MySQL CDR records
I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? Thanks... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Presentation Restricted bit honored?
Hi. I'm wondering if it is possible to make asterisk honor the Presentation Restricted bit on incoming PRI calls. Ideally I'd still like to see the number in the CDR but we can't let users hear restricted numbers in their voicemail messages, etc. The docs only seem to talk about outgoing calls. Thanks... ___ --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] DTMF heard at end of AGI Record File
When I use AGI's Record File with DTMF termination, you can hear a snippet of DTMF at the end of the message. Actually, you hear DTMF at the beginning too, but I work around that by streaming from a few hundred milliseconds into the file. However, I haven't been able to come up with a way around hearing it at the end, and the customer is complaining. If there's no workaround, I'll file a bug report. You can hear an example of this on +1 619 364 0221. Thanks. ___ --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] system() app changed drastically! How do I use it now?
We upgraded to the latest version of asterisk (because we needed some
newer features), only to find all our PIN applications accepting any
number the caller makes up!
I traced this to the System application completely changing the way it
deals with success or failure of the program it calls.
Previously, if the PIN was completely bogus, we exited with -1, which
caused asterisk to jump to priority n + 101 and we told the caller to
take a hike. Now, instead it sets $SYSTEMSTATUS to either SUCCESS or
FAILURE.
But since (as far as I know, without using AEL) there is no conditional
branching based on a variable, how am I supposed to use this?
I'd appreciate any ideas.
Thank you.
Here's an example of our one-time PIN setup.
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Read(PIN,87)
exten = s,4,System(/usr/local/bin/pin -c ${PIN}) ; check it
exten = s,5,System(/usr/local/bin/pin -d ${PIN}) ; delete it
exten = s,6,SetAccount(${PIN})
exten = s,7,Newt,pinout-config ; connect them
exten = s,105,Playback(5021); tell them their PIN is invalid
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Re: [Asterisk-Users] system() app changed drastically! How do I use it now?
On 2005-09-26 at 18:15, Jim Gottlieb ([EMAIL PROTECTED]) wrote:
But since (as far as I know, without using AEL) there is no conditional
branching based on a variable, how am I supposed to use this?
OK, I forgot about GotoIf. However, the doc is wrong (or at least
incomplete), because it only mentions SUCCESS and FAILURE, but I'm
finding SYSTEMSTATUS set to APPERROR.
So I'm doing:
exten = s,5,GotoIf($[${SYSTEMSTATUS} = APPERROR]?105:6)
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[Asterisk-Users] StripMSD or extension parser bug?
For years we've had the following simple context for outgoing calls:
[outtrunk]
; match any NANP, and strip leading 1 off
exten = _1XX,1,StripMSD,1
; dial outbound on trunk group 1
exten = _XX,2,Dial,Zap/g1/${EXTEN}
But when I upgraded on Friday to the latest CVSHEAD, this no longer
works. If I send 13115552368 to this context, I get a message like
pbx.c: Channel 'Zap/361-1' sent into invalid extension '3115552368' in
context 'outtrunk', but no invalid handler
I tried adding a separate line to match 10D:
exten = _XX,1,Dial,Zap/g1/${EXTEN}
but the same call generated a timeout.
I don't know if this is a bug in StripMSD, extension parsing, or user
error.
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[Asterisk-Users] compute traffic intensity from CDR?
Hi. Has anyone written anything that can take CDR output and calculate traffic intensity? We're interested in figuring out the maximum number of simultaneous calls we were handling for various phone numbers / services. Thanks... ___ --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] Caller ID for auto outgoing calls
Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various approximations): Channel: Local/[EMAIL PROTECTED] Callerid: 6193647100 MaxRetries: 5 RetryTime: 60 WaitTime: 60 Context: outtrunk Extension: 16193647100 Priority: 1 SetVar: CALLERIDNUM=6193644799 When it calls 6193644799, it properly shows a Caller ID of 6193647100. But then when it dials 6193647100, it still shows Caller ID of 6193647100 instead of 6193644799. What am I doing wrong? How do I get the Caller ID set correctly for the second half of the call? I've tried various other variables but I haven't been able to get anything to work. Thanks. ___ --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] time includes
If I'm doing a time include in extensions.conf, do I want 04:00-23:00 and 23:00-04:00 or 04:00-22:59 amd 23:00-03:59? I want to make sure that at no time are both or neither included. In other words, does the second time go to HH:MM:00 or HH:MM:59? Thanks. ___ 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] Level 3 SIP -- asterisk
Hi. Can anyone point me to some docs detailing how to set up a connection with Level 3 Communications? A customer of ours wants us to terminate some inbound service via Level 3 to our asterisk server. I've tried all sorts of settings but nothing yet has worked. SIP debug shows a 407 Proxy Authentication Required error. I haven't been able to find anything on the web, and the techs at Level 3 say they've never heard of asterisk and have no idea what I'm doing wrong. Any help would be appreciated. Thanks... ___ 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] what replaced app_qcall?
I see that app_qcall has been replaced. We rely on this for some of our applications. What has it been replaced by? It was nice to be able to just dump files into /var/spool/asterisk/qcall and have the calls be placed automatically. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 191st simultaneous call fails
On 2004-12-17 at 12:13, Vitaly Nikolaev ([EMAIL PROTECTED]) wrote: Have you analized quality of the calls ? what was quality of 190 call ? :) Quality was perfect and a load average of only about 2. ___ 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] 191st simultaneous call fails
I've been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed to create update thread! Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9 Dec 14 15:44:00 WARNING[1215]: Call specified, but not found? Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9 It's not tied to which channel the call comes in on. It's some resource that's exhausted after 190 calls. A limit on threads? I thought it might be per-process file descriptors even though we were only going up to 529 on that PID and I used 'ulimit -n' to increase it before starting asterisk, but that didn't make a difference. # cat /proc/sys/kernel/threads-max 14336 I would think that's enough, but perhaps the per-process limit is much lower. Any clues? Thanks... ___ 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] high-capacity systems / trouble with Tyan
On 2004-10-29 at 20:49, Chris A. Icide ([EMAIL PROTECTED]) wrote: The culprit is the RedHat kernel. I don't know what redhat does with their kernel or sources. But If you build your own kernel from non-redhat source, asterisk will compile perfectly. I did as instructed and recompiled a kernel from kernel.org and rebuilt asterisk. However, the problem remains. I can run one or two cards with no problem. But once I enable the third card, the system locks up within a few minutes. I tried getting Athlon MP motherboards other than the Tyan S2466, but no one has any anymore. ___ 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] high-capacity systems / trouble with Tyan
Hi. We're looking for a reliable platform to run 12 or more T1s in a single system. Very little or no transcoding. Mostly IVR and some conferencing. We have been been running 12 spans using Dual Athlon systems on an older Tyan motherboard and 1500 MP CPUs. This works for about 9 or 10 spans before trouble starts. We wanted to try the new AMD MP 2800 chips on the newer Tyan S2466 motherboard, but the systems hang or panic (with DMA errors) after starting the zaptel drivers. We tried putting the older slower CPUs on the new motherboard and had the same trouble. We've tried to find other Athlon MP motherboards, but no one seems to have any non-Tyan boards in stock. I know there are no simple answers to questions like this. But we'd like to find a platform that can support as many spans as possible. Any suggestions? Should we try dual Opteron? Dual Xeon? Anyone have a source for non-Tyan dual Athlon motherboards? Thanks... ___ 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] high-capacity systems / trouble with Tyan
On 2004-10-29 at 17:02, Chris A. Icide ([EMAIL PROTECTED]) wrote: I currently have a development system I use when developing configurations for my clients. It's a Tyan 2466 motherboard with the latest bios revision, running with two AMD 3000 MP processors. I forgot to mention that the Tyan board worked for us with one T400P. We only started to have trouble when we installed multiple boards. I have had X100P, TDM4XX, and TE4 cards in it with no issue. Have you had multiple cards in it at the same time? I never even tried the 2.4 kernels in the system, I built the 2.6 kernel before installing asterisk. We've been sticking with the 2.4 kernel in Fedora Core 1. I installed FC2 with its 2.6 kernel and couldn't even get asterisk to compile. I realize that Redhat isn't the only Linux. I've only been using it because it's what we've always used and it's what I'm most familiar with (though my girlfriend prefers SUSE). ___ 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] torisa startup troubles
The hard disk on an old system with an ISA card just died and I reloaded a more modern OS (Fedora Core 1) and asterisk, but I wonder what I need to do to get ISA support working. # /etc/init.d/zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcusb Running ztcfg: ZT_SPANCONFIG failed on span 1: No such device or address (6) [FAILED] # grep torisa /etc/mod*conf /etc/modprobe.conf:options torisa base=0xd --same as before /etc/modprobe.conf:alias char-major-196 torisa /etc/modprobe.conf:post-install torisa /sbin/ztcfg If I try a manual modprobe: # modprobe torisa /lib/modules/2.4.22-1.2115.nptl/misc/torisa.o: init_module: Input/output error Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.22-1.2115.nptl/misc/torisa.o: insmod /lib/modules/2.4.22-1.2115.nptl/misc/torisa.o failed /lib/modules/2.4.22-1.2115.nptl/misc/torisa.o: insmod torisa failed Any ideas? Thanks... ___ 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] panic() panic() panic() and dma errors
On 2004-06-25 at 22:12, Steve Hanselman ([EMAIL PROTECTED]) wrote: If you cat /proc/interrupts is anything else sharing with the TEs? It doesn't seem to: CPU0 CPU1 0: 5413 5623IO-APIC-edge timer 1: 0 5IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 14: 2722 3918IO-APIC-edge ide0 15: 19 19IO-APIC-edge ide1 16: 45283 32056 IO-APIC-level tor2 17: 31891 45048 IO-APIC-level tor2 18: 31625 44915 IO-APIC-level tor2 19:176 3 IO-APIC-level eth0 NMI: 0 0 LOC: 10947 10945 ERR: 0 MIS: 0 I've turned off everything in the BIOS that I can like all serial ports, parallel port, APCI. Interestingly, the system no longer dies in a panic() but with DMA errors scrolling across the console. hda: DMA interrupt memory hda: lost interrupt hda: dma_timer_expiry: dma status == 0x24 If I pull out the T400P boards, no problems. I'll leave my original message in below as it's been a while. I've been away most of the summer and I leave for Asia on Sunday but I'm trying to resolve this as best I can. Thanks. -Original Message- From: Jim Gottlieb [mailto:[EMAIL PROTECTED] Sent: 25 June 2004 20:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] panic() panic() panic() Hi all. I've been trying to build some new systems, and no matter what I do, if I load the zaptel and tor2 drivers, the system panics within an hour, even with no traffic. These systems are using dual Athlon MP 2800 chips with one, two, or three T400P boards and 2 GB of system memory. I'm currently using Fedora Core 1, but I also went back to our old reliable Red Hat 7.3 and the systems still panic()ed. If I don't start the zaptel driver, they don't panic. If I start the zaptel driver, but don't start asterisk, they still panic. I'm at a loss of what to try next. A typical Call Trace from the panic message looks like: wait_on_irq, [kernel] 0xde __global_cli [kernel] 0x62 flush_to_ldisc [kernel] 0x126 __run_task_queue [kernel] 0x61 context_thread [kernel] 0x13b context_thread [kernel] 0x0 context_thread [kernel] 0x0 kernel_thread_helper 0x5 Any ideas? Thanks... ___ 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] panic() panic() panic()
Hi all. I've been trying to build some new systems, and no matter what I do, if I load the zaptel and tor2 drivers, the system panics within an hour, even with no traffic. These systems are using dual Athlon MP 2800 chips with one, two, or three T400P boards and 2 GB of system memory. I'm currently using Fedora Core 1, but I also went back to our old reliable Red Hat 7.3 and the systems still panic()ed. If I don't start the zaptel driver, they don't panic. If I start the zaptel driver, but don't start asterisk, they still panic. I'm at a loss of what to try next. A typical Call Trace from the panic message looks like: wait_on_irq, [kernel] 0xde __global_cli [kernel] 0x62 flush_to_ldisc [kernel] 0x126 __run_task_queue [kernel] 0x61 context_thread [kernel] 0x13b context_thread [kernel] 0x0 context_thread [kernel] 0x0 kernel_thread_helper 0x5 Any ideas? Thanks... ___ 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] IAX2 call causes SEGFAULT
Hi. I'm trying to do a pretty generic IAX2 call between two asterisk machines, but when the call arrives, I get a SEGFAULT. The receiving machine is running the latest code from the stable branch, though this also happened with a snapshot from 2004-01-30 so I don' think it's a recent problem in the code. More likely something I'm doing wrong, but I can't figure out what. The IAX2 debug shows: x-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 1 DCall: 0 [192.168.14.21:4569] VERSION : 2 CALLED NUMBER : 16026247788 CALLING NUMBER : 602624 LANGUAGE: en USERNAME: guest FORMAT : 2 CAPABILITY : 65282 ADSICPE : 2 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 1ms SCall: 1 DCall: 1 [192.168.14.21:4569] FORMAT : 2 tus01*CLI Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 000 Type: VOICE Subclass: 2 Timestamp: 00019ms SCall: 1 DCall: 0 [192.168.14.21:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 1 DCall: 1 [192.168.14.21:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: (11?) Timestamp: 2ms SCall: 1 DCall: 1 [192.168.14.21:4569] tus01*CLI Disconnected from Asterisk server The backtraces seem to show something different each time. Once it referenced chan_iax2.c, but other times it seems to show random stuff: #0 0x4018f90e in __select () from /lib/i686/libc.so.6 #1 0x462514cc in ?? () #2 0x0805c723 in ast_waitfor (c=0x80ef2c0, ms=1000) at channel.c:930 #0 0x4018e227 in __poll (fds=0x80ede90, nfds=1, timeout=-1) at ../sysdeps/unix/sysv/linux/poll.c:63 #1 0x08051a8c in ast_io_wait (ioc=0x80ede70, howlong=-1) at io.c:254 #2 0x42542ef7 in do_monitor (data=0x0) at chan_mgcp.c:2543 #3 0x40027bfd in pthread_start_thread (arg=0x42749c00) at manager.c:262 Thanks... ___ 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] tor2 driver panics with 2 sticks of memory
We use dual Athlon machines with up to three T400P 4-span T1 cards. If I have more than one stick of memory (2 1GB modules or 2 512K modules, each identical), I'm getting a panic soon after I modprobe the tor2 driver. I just loaded the latest from CVS and I'm still getting the panics, which look in part like: Apr 16 14:42:28 test71 kernel: wait_on_irq, CPU 0: Apr 16 14:42:44 test71 kernel: irq: 1 [ 0 1 ] Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] Apr 16 14:42:47 test71 kernel: Stack dumps: Apr 16 14:42:47 test71 kernel: CPU 1: 000 0 Apr 16 14:42:47 test71 kernel: 00 00 Apr 16 14:42:47 test71 kernel: 00 00 Apr 16 14:42:47 test71 kernel: Call Trace: [f894d3a0] ohci_hcd_list [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: [f894d3a0] ohci_hcd_list [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: [f894ac60] rh_int_timer_do [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 kernel: CPU 0:f6a2bea4 c023f901 0001 fff f c010a362 c023f916 Apr 16 14:42:47 test71 kernel: f79ce6a4 f6a2bef8 c017f574 04 00 0005 04bf 8a31 Apr 16 14:42:47 test71 kernel:7f1c0300 01000415 1a131100 170f1200 00 00 f6a2a000 f782d978 f782d978 Apr 16 14:42:47 test71 kernel: Call Trace: [c010a362] __global_cli [kernel] 0x e2 Apr 16 14:42:47 test71 kernel: [c017f574] change_termios [kernel] 0x24 Apr 16 14:42:47 test71 kernel: [c017f844] set_termios [kernel] 0x164 Apr 16 14:42:47 test71 kernel: [c017c6e2] tty_ioctl [kernel] 0x352 Apr 16 14:42:47 test71 kernel: [c0151887] sys_ioctl [kernel] 0x257 Apr 16 14:42:47 test71 kernel: [c0108c5b] system_call [kernel] 0x33 Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 last message repeated 2 times Apr 16 14:42:47 test71 kernel: wait_on_irq, CPU 0: Apr 16 14:42:47 test71 kernel: irq: 1 [ 0 1 ] Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] Apr 16 14:42:47 test71 kernel: Stack dumps: Apr 16 14:42:47 test71 kernel: CPU 1: 42029098 000 [...] Any ideas? Thanks... ___ 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] regexp problems
I'm trying to filter calls that don't have a proper ANI. This is what I did: ; only if they a real-looking ANI exten=_1XX1118/_.N.,1,Newt,1118-config ; Otherwise, send them to the loser partyline exten=_1XX1118,1,Goto(outtrunk,19096611234,1) This properly deals with null ANIs, but for some reason those with ten zeroes get matched by the first line. I also tried to be a bit more specific, like: exten=_1XX1118/_.[1-9][1-9].,1,Newt,1118-config but that also matched on all zeroes. Am I doing something wrong or is this a bug? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On 2003-07-02 at 13:54, Steven Critchfield ([EMAIL PROTECTED]) wrote: Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones Unfortunately, not all fax machines send the CNG tone, so using a separate fax number with distinctive ring is far more reliable. Can asterisk detect the various rings and route accordingly? If not, he could get two TDM100B cards, plug the two outputs of his distinctive ring detector into the two cards and route each one to a different context. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Video changes
On 2003-07-01 at 16:00, Tamas Levente ([EMAIL PROTECTED]) wrote: Seems like windows messenger is using it for video comm Likewise, I notice that Apple's new iChat AV requires that port 5060 is open, so it looks like they too are using SIP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FGB not waiting for digits
On 2003-07-01 at 13:27, Jim Gottlieb (That's Me!) wrote: -- Starting simple switch on 'Zap/1-1' == Unknown extension 's' in context 'intrunk' requested I also see logged: File chan_zap.c, Line 3833 (ss_thread): Got a non-Feature Group B input on channel 1. Assuming EM Wink instead Looking at that code, it seems to be dealing with the digits received (since the only thing that distinguishes FG B is the digit string), but I get that message immediately, before I even have the chance to send any digits. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Newbie Question
On 2003-06-27 at 14:24, Chip Mefford ([EMAIL PROTECTED]) wrote: Is anyone actually using * as a primary phone system in a small/medium sized business with more than a dozen stations and a real receptionist who handles calls? As impressed as I am with asterisk, and as happy as we are with it as the basis for our IVR/conferencing application, I don't think it is ready to replace a real PBX for general office use. And it doesn't have to because they can work together. There are a lot of very reasonably priced systems on the used market. For example, we use an Eon Millennium (née ITT 3100) that we picked up fully loaded for a few thousand dollars, and for VoIP/IVR/ACD/VM we connect to an asterisk server through its PRI interface. But the PBX itself provides the standard features like nice feature phones (available refurbished for one-third the price of a Cisco 7960), busy lamp / DSS consoles, and ARS tables, that are nicer than anything you could cobble together easily with asterisk at this point. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is zap destroy channel safe?
I have used zap destroy channel a few times to hang up on a caller, but each time, things seemed to go badly wrong soon after. The first time, I could no longer connect to the asterisk console, and this latest time, the program segfaulted a few minutes later. Is this zap destroy channel not something that should be used on a production system? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and DISA fixes
It would be really nice if the DISA allowed one to make follow-on calls by pressing some key sequence (say, press * three times in a row within one second). This especially helps when you are at a hotel that charges for each call. We put our DISA on a toll-free number, and as long as it supports follow-on calls, you can make calls all night for that $1 they charge per call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI matching trouble
On 2003-05-28 at 22:39, Mark Spencer ([EMAIL PROTECTED]) wrote: exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue Take out the _. rule and just leave it 4044633 and it should work fine. That did it. Works great! Thanks. Not postive the _ is required on matching the callerid part, but honestly i just don't remember. It _is_ required. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ANI matching trouble
Hi. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,1,OurApp,sandiego-queue exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue but it didn't seem to work. Every call went to the default queue. I also tried exten = 4044633/_619XXX,1,OurApp,sandiego-queue to no avail. It did work if I put a specific number in there: exten = 4044633/6193644788,1,OurApp,sandiego-queue but of course I can't list every possible number. What am I doing wrong? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Up coming FXS card.. but what about FX0?
On 2003-03-29 at 12:29, Gary [EMAIL PROTECTED] wrote: Consider a card which could be configured as selectable fxo or fxs and 12 to 20 ports. Sounds to me like a channel bank. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error message: unable to set audio mode
I just upgraded to the latest CVS and started noticing that I'm getting error messages every time a call hangs up: WARNING[598031]: File chan_zap.c, Line 1879 (zt_setoption): Unable to set audio mode on channel 37 We are also seeing this for every call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some problems about X100P in Japan
On 2003-03-25 at 14:46, Shinsuke,Iwata ([EMAIL PROTECTED]) wrote: I'm having problems with Asterisk and Digium's X100P in Japan. Asterisk doesn't detect PSTN hangup. I'm trying to use Japan and Zero voltage ring options in wcfxo.c, like this #define JAPAN #define ZERO_BATT_RING You shouldn't need ZERO_BATT_RING unless you're using a line coming off of a TA, and if you are then you may not get disconnect supervision, depending on your TA. In any case, the disconnect supervision (AKA CPC) on PSTN lines in Japan is very very short. Look in wcfxo.c and play with the value of BATT_DEBOUNCE. Try lowering it until you get hangup detection. You may also want to verify that you _are_ receiving the CPC signal. Either use an analog multimeter across the line or hold down the tone on a line-powered tone phone while the calling party hangs up and see if the tone or voltage goes away, even very briefly. Or use INSnet 1500 :-). --- Jim Gottlieb (IRC: TokyoJimu) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX weirdness
I loaded the latest CVS this morning on the machine we use as our main IAX gateway. I often use this gateway to connect to an IVR machine that doesn't have a T-span, so I connect via IAX. However, after installing the new * release, calls to the IVR machine acted strangely. The main problem was that various voice prompts were missing or would cut off after a brief instant. For example, instead of playing Please record your message at the tone, beep, it just played the beep. There was no dead time when it should have been playing the prompt; it just skipped it. (Our prompts are in µ-law format if that makes a difference.) When I pressed the option for billing to a credit card, instead of Enter your credit card number, it would just play Ent This was not completely consistent. On some calls the prompts would play; on others we would get this behavior. Reverting to our prior * version on our IAX gateway made everything work again. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA-186 and fake ring
I am using an ATA-186 connecting to an asterisk SIP gateway. When I dial out through it (via a PRI) to a real number, I notice that I hear a fake ringback tone. For example, if I call my voicemail, which answers without a ring, I still hear a bit of ringback when I call via SIP. In fact, if I called a busy number, I never heard a busy. Just continuous ringback, as if it's just playing me local ringback until it sees answer supervision, at which time it cuts the call through. I alleviated this by adding a line: exten=_XX,3,Busy so now it goes to busy when the number I call is busy, but, actually, I still hear a ringback tone first, and then it goes to busy. Who is generating this ringback? The ATA or asterisk? What if I call a non-suping number with a the number has been changed recording? Will I never hear it because audio will never be cut through without answer supervision? The relevant lines from my extensions.conf: ; match any US, and strip leading 1 off exten=_1XX,1,StripMSD,1 ; dial outbound on trunk group 1 exten=_XX,2,Dial,Tor/g1/BYEXTENSION ; if we don't put this in, we'll hear ringback forever on a busy number exten=_XX,3,Busy Thanks for putting up with this relatively green asterisk user... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 and fake ring
On 2003-03-12 at 09:44, you wrote: Who is generating this ringback? The ATA or asterisk? Find out by doing a trace. If you're using callprogress, then you should see a 180 Ringing sent to the ATA when we detect ringing on the FXO. If you're not using call progress, then we should not be sending 180 ringing. We're not using callprogress (at least it's not set in zapata.conf). I also tried explicitly setting it to 'no', reloading, and trying again. No change. I do get a 180 Ringing. I am dialing from my ATA-186 to 18189950699, a telco busy test. Yet all I hear is a ring, though on one call I heard a short blip of busy before the ring started. [See attachment for the trace output] You can also use the new debug channel Zap/1-1 to see if the FXO is ringing. If I do a 'show channel' on it, I get: Zap/21-1 (intrunk6197474525 1 ) Ringing AppDial (Outgoing Line) SIP/0054-b2bc (outtrunk 8189950699 2 )Ring Dial Tor/g1/BYEXTENSION That 6197474525 is strange. That's probably the DNIS used on the last incoming call to that channel, but has nothing to do with my outgoing call. I was calling from 6193640054 (SIP 0054). My definition in sip.conf is: [0054] type=friend insecure=yes secret=myownsecret callerid=Jim Gottlieb (619) 364-0054 ; dynamic binding seems to time-out; try defaultip host=dynamic defaultip=192.168.40.90 ; need to set the following so we can use voicemail and other DTMF apps dtmfmode=rfc2833 [I dialed...] Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=850095511 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA v2.15 ata18x (020927a) Expires: 300 Content-Length: 247 Content-Type: application/sdp v=0 o=0054 5680 5680 IN IP4 192.168.40.90 s=ATA186 Call c=IN IP4 192.168.40.90 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Interface is eth0 IP Address is 198.51.175.9 Using latest request as basis request Sending to 192.168.40.90 : 5060 (non-NAT) Capabilities: us - 14, them - 13, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.40.90:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=850095511 To: sip:[EMAIL PROTECTED];user=phone;tag=06e4b177 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED];user=phone Proxy-Authenticate: Digest realm=asterisk, nonce=028f4554 Content-Length: 0 to 192.168.40.90:5060 Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=850095511 To: sip:[EMAIL PROTECTED];user=phone;tag=06e4b177 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=850095511 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username=0054,realm=asterisk,nonce=028f4554,ur i=sip:[EMAIL PROTECTED],response=a8dbe8f8d6faee139756514c82cad48f Expires: 300 Content-Length: 247 Content-Type: application/sdp v=0 o=0054 5686 5686 IN IP4 192.168.40.90 s=ATA186 Call c=IN IP4 192.168.40.90 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.40.90 : 5060 (non-NAT) Capabilities: us - 14, them - 13, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 18189950699 in sip Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.40.90:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=850095511 To: sip:[EMAIL PROTECTED];user=phone;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED];user=phone Content-Length: 0 to 192.168.40.90:5060 We're at 198.51.175.9 port 53614 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.40.90:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=850095511 To: sip:[EMAIL PROTECTED];user=phone;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED];user=phone Content-Type: application/sdp Content-Length: 211 v=0 o=root 5860 5860 IN IP4 198.51.175.9 s=session c
Re: [Asterisk-Users] Known SIP - NAT Solutions?
On 2003-03-05 at 15:08, you wrote: I would like to put these SIP phones into are behind NAT. I was quite surprised when I tested Vonage's service that I could plug in their ATA 186 behind my NAT/VPN box and it immediately worked, even for incoming calls. They must make an outbound connection and hold it open for the incoming calls that occur. Assuming this is standard SIP stuff, I wonder if Asterisk does or could support this type of registration/setup. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A new pbx for an office in the UK
On 2003-02-26 at 16:37, you wrote: is it possible to mix analog and digital (ISDN) handsets on this TSU600 channel bank? No. A channel bank's output is all analog. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A new pbx for an office in the UK
On 2003-02-26 at 14:38, Steven Critchfield [EMAIL PROTECTED] wrote: I'll admit I have not been in a company where there was heavy phone usage, but I've not seen anyone use more than a few keys outside of the stock DTMF keypad. The only thing I know for sure that isn't emulated is the intercom function. Some things I like to have on my desk phone, that can not easily be emulated with speed-dial buttons: · Busy lamps, so I can see who is on the phone before I decide to disturb them · Multiple virtual line appearances · Message waiting lamp · Display that shows who is calling whom in my pickup group so I can decide which calls to grab · Quick single button access to conferencing, transfer, and party drop ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A new pbx for an office in the UK
On 2003-02-26 at 16:26, you wrote: display a list of outside lines and their status (in use to whom, on hold, or available), and if on hold the amount of time they have been that way, and allow pickup We actually have this, as we use an in-house app that runs under asterisk for our call center agents. Some day I hope to make the case to management to release some of this stuff. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
