Re: [asterisk-users] Dial C option
When I use
exten = _70XX,1,NoCDR()
exten = _70XX,2,Dial(SIP/${EXTEN}|20|tr)
I get
Executing NoCDR(SIP/7002-081ac898, ) in new stack
Aug 28 15:27:18 WARNING[4670]: cdr.c:443 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' not posted
Aug 28 15:27:18 WARNING[4670]: cdr.c:445 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' lacks end
-- Executing Dial(SIP/7002-081ac898, SIP/7003|20|tr) in new stack
I am using 1.2.11
Regards
Moises Silva wrote:
We use our own CDR, but as I understand, the C option resets the CDR,
that does not means is not going to save cdr, but is going to restart
the CDR. So, a simple NoCDR() before dialing should work, or ForkCDR()
and then NoCDR() if you want to save previous data.
Regards
On 8/27/06, Master Abi [EMAIL PROTECTED] wrote:
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial
line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
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Re: [asterisk-users] Dial C option
That is what I thought, but then how do I STOP recording CDR's. If I use
it in the h extension, it also gives a warning.
Moises Silva wrote:
Normal behaviour since the call record before executing NoCDR() was
not posted (saved)
Regards
On 8/28/06, Master Abi [EMAIL PROTECTED] wrote:
When I use
exten = _70XX,1,NoCDR()
exten = _70XX,2,Dial(SIP/${EXTEN}|20|tr)
I get
Executing NoCDR(SIP/7002-081ac898, ) in new stack
Aug 28 15:27:18 WARNING[4670]: cdr.c:443 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' not posted
Aug 28 15:27:18 WARNING[4670]: cdr.c:445 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' lacks end
-- Executing Dial(SIP/7002-081ac898, SIP/7003|20|tr) in new
stack
I am using 1.2.11
Regards
Moises Silva wrote:
We use our own CDR, but as I understand, the C option resets the CDR,
that does not means is not going to save cdr, but is going to restart
the CDR. So, a simple NoCDR() before dialing should work, or ForkCDR()
and then NoCDR() if you want to save previous data.
Regards
On 8/27/06, Master Abi [EMAIL PROTECTED] wrote:
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial
line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
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[asterisk-users] Dial C option
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
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[Asterisk-Users] HT-1000 chipset experience
Hi I am about the purchase a server and would like to know if anyone has had any experience with the TE410P Rev 2 in a server that has a ServerWorks BCM5785 (HT-1000) chipset. Thanks Master ___ --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] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!
Cory, An easier way to do it: (used gentoo) 1. Connect a PATA drive and install gentoo with 2.6.14 and include Marvell SATA driver. 2. Use Ghost for Linux V.0.17 and copy PATA disk to SATA disk. 3. Disconnect the PATA 4. Boot from the install CD and change grub.conf and fstab 5. Reboot and be happy Another way is to build your own universal CD. This way you do not have fiddle with drives. Master David Muench wrote: On 12/21/05, Cory Andrews [EMAIL PROTECTED] wrote: I have a SuperMicro 5013C-MT with the P4SCT+ motherboard and am having trouble with all Linux distributions (Debian, Gentoo, Redhat ES3 and Ubuntu). No distributions will detect the SATA drives and therefore cannot install. Hi Cory, I have that system as well - excellent system, but it was frustrating getting Linux on it. Here's what I did with Ubuntu: Boot up the Ubuntu live cd. apt-get build-essential and the kernel sources. Download the marvell SATA driver. I am using 3.4.2a, and have been for 6+ months with no issues. If you're using Ubuntu 5.10 which has kernel 2.6.12, you'll need a patch to get the driver to compile - send me an email directly if you can't find it on google. Build the driver in the livecd, and then copy mvSata.ko off to another machine. Then boot the Ubuntu install CD, and after it sets up the network but before it gets to the partitioning, ALT-F2 into a shell and grab that mvSata.ko from the machine you copied it to. modprobe that in the Ubuntu shell and the disks should be available. You should be able to proceed through the Ubuntu install now. The next problem is that Ubuntu has no knowledge of that mvSata driver, so it won't be part of the initrd once you finish the install and reboot, so Ubuntu won't boot. Boot up the livecd again, grab the mvSata.ko off of your other machine to get the disks online, and then generate a new initrd in your ubuntu install. Basically you need to copy mvSata.ko to /lib/modules/kernel ver/kernel/drivers/scsi/ and then do a depmod with the -b option since your real root partition will be mounted somewhere else like /mnt or wherever you mounted it. After that use mkinitrd to generate a new initrd including the mvSata module. This sounds like a heck of a lot of work but it's not so bad. Once you get it installed once, kernel upgrades are easy - you just need to put the mvSata in place and regenerate the initrd after installing the new kernel. If you have any questions, feel free to ask. Dave -- David Muench - [EMAIL PROTECTED] Jabber ID: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning CONFIG_ZAPATA_DEBUG on 2.6.14
Hi Upgraded to Gentoo 2.6.14-r2. When compiling zaptel, warning appears. Zaptel module loads fine. Cannot remember seeing this on 2.6.13. Is there another Kernel switch that needs to set. CRC and RTC is set in kernel. make[1]: Entering directory `/usr/src/linux-2.6.14-gentoo-r2' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:1736:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:1923:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3032:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3039:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3048:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3295:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5287:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5806:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5876:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5899:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:176: warning: 'fcstab' defined but not used Master ___ --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] TE405P V2 changes?
Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ 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] TE405P V2 changes?
Are you using Redhat/Fedora? If I remember those init scripts is for Redhat/Fedora. I am using gentoo. Did you make any modifications to wct4xxp.c. or pass any parameters to zaptel. I see there is a #define SUPPORT_GEN1 in to wct4xxp.c which I commented out, but it made no difference. ztcfg seems to where the channels become unassigned. Thanks again. Kib Eki wrote: Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zap to zap bridging not hanging up
Hi
I am trying to develop a night divert. Caller dials in after hours on
Zap and it gets divert to a mobile number via a second Zap. The call
bridges but will not hangup the channels when the parties finish.
Is there something I am missing or an dial option that I should be
using. I am using latest CVS.
[night]
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Set(TIMEOUT(digit)=3)
exten = s,4,Set(TIMEOUT(response)=6)
exten = s,5,Set(dvt=${DB(DIVERT/MOBILE)})
exten = s,6,Gotoif($[${dvt} != ]?s|7:s|103)
exten = s,7,Dial,${PSTNTRUNK}/${dvt}|30|tr
exten = s,8,Hangup
[default]
include = melton-night|17:31-8:59|mon-fri|*|*
Thanks
master
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[Asterisk-Users] Dial W option usage
Hi all Could someone please care to share an example of the Dial W option usage. I cannot seem to find any reference to it usage. I know you use *1 in features.conf to start the monitor, but from there I am lost. Master ___ 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] Sipura SPA-841 Phone Review
if I have conf = 80,111 in meetme.conf, I dial 80# and connect to the conference, then I dial 111#, it indicates pin is incorrect. with other phones it works. Is there something special in the sipura config that will allow more digits after the # master Craig wrote: I found the speaker phone and the headset work ok on the original v.9.x software that came with the units, when I upgraded 2 of them to v 3.x the headset and speakerphone become unusable. I am looking to try and downgrade these units back to v 0.9 so I can use the headset on them. It would be nice to use share call appearances with * so I can turn them into a key telephone system like the system they replaced, but that is something I will have to work on. Apart from that they are brilliant for the price craig Date: Tue, 19 Apr 2005 12:36:09 -0400 (EDT) From: Paul Dugas [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura SPA-841 Phone Review To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;charset=iso-8859-1 On Tue, April 19, 2005 10:05 am, Me said: If Sipura could make the headset jack solid, it would be a great, affordable phone in my opinion. Never had a problem with the headset jack. Now the speakerphone... They ought to be ashamed of themselves for advertising it as a feature of the unit. It absolutely stinks. Totaly useless. Also, very little in response to repeated request for attention on a fix other than try the latest firmware which does little other than making it even worse. Criminal! Paul ___ 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] OT: VIA Mini-ITX, Asterisk, and hardware
I use the MII 1.2Ghz version with TE110P. No problems. Can do about 8-10 ulaw to GSM, possibly more. Also used TDM400 that works fine. Note the MII 1.2 version cannot boot off the CF unless you use FreeBios. Use the EPIA MS version to boot from onboard CF. C. Tomlinson wrote: Hi, I run * on the first 800mhz version they released. I do not use any PCI cards, so cannot coment on that I'm afraid. It works fine for testing in the environment I use...but I haven't stressed it at all. I had to make a change to the makefile for the processor, but I doubt that is needed for the newer versions. If I am right you made the cool little CF + flash disk * distro? I think they are an ideal pair. One of the new mini-itx boards comes with compact flash onboard, and has no builtin sockets except LAN and VGA. Very easy to make an embedded system. I think any of them 266 geode! C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: 20 March 2005 21:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OT: VIA Mini-ITX, Asterisk, and hardware Hello everyone, Does anyone out there have actual experience with running * on a mini-itx board from VIA? They look good, but I have some reserves because of VIA's problems with PCI latency in recent years (audio dropouts, wierd things happening). I am looking at the EPIA CL-1. For $270, I can get a CL-1 (1ghz C3, dual ethernets, etc), 256mb RAM, and a nice small (12 x 2 x 11) case (with 1 4cm fan)... They look like a good next step (or leap) up from a Soekris Net4801. I know that 1ghz C3 != 1ghz intel, but it's still probably better than a 266mhz Geode... I would love to try this board with Sangoma A101's, te110p's, and even some TDM4xx's, but if people out there already know that * is a bad fit here, I probably won't even bother and look elsewhere. Any tips, notes, caveats, etc from anyone? Anyone using any of the hardware I mentioned with one of these boards? Thanks! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 841 issues
Grandstreams do, Sayson 480i does, so does all softphones. They should, because how are you going to se what you typing. Not having a backlit display is bad design. C F wrote: I haven't seen a sip phone that once connected will show the digits pressed on the screen. My SPA 841 doesn't give me any backlit on the display. So I think that not. On Sun, 13 Mar 2005 23:31:03 +1100, Master Abi [EMAIL PROTECTED] wrote: Hi Just 2 issues I have with SPA841. 1. I autodial extension 600 then inside an AGI wait for more digits. The digits are transmitted correctly to * but they do not show up on the SPA841 display, only the 600. How do I set the 841 is show the digits after the 600# 2. Is the SPA841 pixel display backlit? Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 841 issues
Hi Just 2 issues I have with SPA841. 1. I autodial extension 600 then inside an AGI wait for more digits. The digits are transmitted correctly to * but they do not show up on the SPA841 display, only the 600. How do I set the 841 is show the digits after the 600# 2. Is the SPA841 pixel display backlit? Master ___ 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] Embedded Asterisk Paper Complete
Could you email me the PDF I am having PASV FTp problems. I have the same setup. Out of interest which case are you using. I looked at the CF adaptor you used, but not sure if the Morex 3677 case I am using is high enough. Kilburn JR Richardson wrote: Hi all, The journey is complete, at least for this project. http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html I spent the better part of Halloween putting this together, I hope its useful, enjoy. My ftp server is on the fritz so feel free to post on any other user sites. If you have any difficulties, email me and Ill send the files to you directly. JR ftp://odyssey-tech.net/Embedded_Asterisk.doc ftp://odyssey-tech.net/Embedded_Asterisk.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream G726-32 now working properly with *
Hi, G726-32 codec from beta firmware 1.0.4.54 now works fine with *. Tested on BT101 and HT286 over a 64K DSL line. Some progress but iLBC still has not surfaced. Get it from http://www.grandstream.com/BETATEST/ Master ___ 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] g726 Codec w/ GrandStream 1.0.4.50 Firmware
Upgrade to the latest CVS and ast_rtp_read/write warnings will disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works great. Master Greg Boehnlein wrote: Hello all, I'm trying to get the g726 codec patch contained in: http://bugs.digium.com/bug_view_page.php?bug_id=0001104 to work with the latest GrandStream beta firmware and I am a lot closer than I was a couple of weeks ago with the 1.0.4.46 firmware. I am now hearing Audio that is distinguishable with the .50 firmware release, but Asterisk is giving me the following error messages on the console: == Spawn extension (default, 8500, 1) exited non-zero on 'SIP/damin-8ebc' -- Executing VoiceMailMain(SIP/damin-e62c, ) in new stack Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 -- Playing 'vm-login' (language 'en') Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples for format G726 Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples for format G726 Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples for format G726 Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 Any suggestions on where I should look? Could this possibly be a configuration issue on my part? ___ 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] g726 Codec w/ GrandStream 1.0.4.50 Firmware
I am not running the V1-0stable. Use the development version. My version is 2 days old. G726 added to development CVS about 10 days ago. Greg Boehnlein wrote: On Mon, 8 Mar 2004, Master Abi wrote: Upgrade to the latest CVS and ast_rtp_read/write warnings will disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works great. Hmm.. when was this fixed? I'm running a CVS version that was pulled and built this morning, however I believe that I'm running the 1.0_stable branch on this box. Let me clean up and rebuild and see if that corrects the issue. Master Greg Boehnlein wrote: Hello all, I'm trying to get the g726 codec patch contained in: http://bugs.digium.com/bug_view_page.php?bug_id=0001104 to work with the latest GrandStream beta firmware and I am a lot closer than I was a couple of weeks ago with the 1.0.4.46 firmware. I am now hearing Audio that is distinguishable with the .50 firmware release, but Asterisk is giving me the following error messages on the console: == Spawn extension (default, 8500, 1) exited non-zero on 'SIP/damin-8ebc' -- Executing VoiceMailMain(SIP/damin-e62c, ) in new stack Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 -- Playing 'vm-login' (language 'en') Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples for format G726 Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples for format G726 Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples for format G726 Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending format G726 packets Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format G726 Any suggestions on where I should look? Could this possibly be a configuration issue on my part? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup to CDR recording timing
Hi What is the relationship between when CDR recording occurs and the hangup extension is executed. Normally CDR happens before the h extension is executed. I use the h extension to clean up for routines, but sometimes it gets called to quickly before the CDR is dumped into a DB. I would like the h extension to execute after CDR recoding. Is there a way to force or is it depend on which party hangups. I also use the g option in Dial but this does not completely solve this issue. Using CVS from Asterisk CVS-02/09/04-20:25:52. Thanks Master ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Need to interface to BRIs
Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ 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] retrans_pkt: Maximum retries exceeded on call
I think this is related to a device (GS in my case) that has an sip entry but you physically removed it and switched it off. Somehow * still thinks connected. Comment out the entry and reload or put the device back. Mark Rizzo wrote: I have seen similar error which coincided with my GS phone taking a call-waiting call while I was on the GS phone. I got two of the errors (101 102 I think) and then the GS phone or Asterisk terminated the call I was on (including the call-waiting call that was trying to get through). I chalked this up to missing configuration setup or that GS does not support call-waiting but had not researched yet. Mark -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Chris Wilson *Sent:* Saturday, January 24, 2004 12:26 AM *To:* [EMAIL PROTECTED] *Subject:* [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.Any help would be appreciated! =] Thanks! Chris ___ 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] SIP + ADPCM
Checked the archives. I cannot get ADPCM to work with SIP. Calling from phone1 (adpcm) to phone 2(ulaw). Both phones Grandstreams with one set with G726-32 with v0.7.1 cvs. Has anyone got adpcm to work? Jan 24 09:00:14 WARNING[409617]: rtp.c:1069 ast_rtp_write: Not sure about sending format ADPCM packets Jan 24 09:00:14 WARNING[409617]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format ADPCM Master ___ 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] ADSI phone vs. IP phone
Aastra will have a production PT480i SIP phone in March for ~US180-$200. Same phone as ADSI model just SIP, but has 4 extra buttons for virtual lines. Got a beta SIP model under test. Designed for SIP v1 v2. * is one of PBX used for testing by development, so should be * friendly when released. Master Tim Thompson wrote: I've been pretty satisfied with the Aastra PT480. There are some other people that say they don't like them, but I think the $110-$120 ea. Works great for our office and the people I install for. Take it for what you paid for it. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 12:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. To quote myself: True, but I don't have to retool my office and install POE switches to use ADSI phones, either. No, I will not put a hub/switch at every desk and then use wall-warts for every phone to get around retooling the office. :-) I'm not going to bastardize my network by placing the equivalent of a 3- port switch or hub at every desk to have the phone system compete with our heavy network users (CAD mostly), and I will fight tooth and nail against having to put a goddamned wall-wart at every station just to power the damned IP phones. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pause after dialed option
I had experienced this problem before. I found this to be related to 2 items. Firstly, try not to use the s,1 starting each submenu. Secondly, if there are more than 20 sub menus, you will get this delay problem. Why I do not know. I reordered and regrouped and the problem disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Carr Sent: Thursday, 13 November 2003 1:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pause after dialed option
Use like this... [mainmenu] exten = s,1,Goto(sales|100|1) exten = s,2,Goto(support|200|1) [sales] exten = 100,1,Answer ; Answer the line exten = 100,2,DigitTimeout,5 ; Maximum Timeout between digits exten = 100,3,ResponseTimeout,10 ; Maximum Timeout awaiting response exten = 100,4,BackGround,mainmenu; Play Main Menu [support] exten = 200,1,Answer ; Answer the line exten = 200,2,DigitTimeout,5 ; Maximum Timeout between digits exten = 200,3,ResponseTimeout,10 ; Maximum Timeout awaiting response exten = 200,4,BackGround,mainmenu; Play Main Menu .. etc, etc Also, I don't think putting digit timeouts are always required, but I did find Answer is a fairly safe bet. Try and use s extension is a minimum. Master -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, 13 November 2003 2:11 PM To: Master Abi Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option So what do you use instead of s,1? My s extensions set things like response timeout, digit timeout, etc. Thanks again. AJ On Thu, 13 Nov 2003, Master Abi wrote: I had experienced this problem before. I found this to be related to 2 items. Firstly, try not to use the s,1 starting each submenu. Secondly, if there are more than 20 sub menus, you will get this delay problem. Why I do not know. I reordered and regrouped and the problem disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Carr Sent: Thursday, 13 November 2003 1:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
10 - A way to lock the phone settings (IP address, etc). It is too easy to change the settings when in a public environment. The MENU button should not be 1 press away from changing the settings, Use MENU + SOME COMBINATION. 7 - Use the conference button to access Meetme. Like the Voice Mail UserID and Offhook Auto-Dial where you can preset an extension. OR call the Button Conference/Queue. 8 - Crank up the speakerphone volume. In a public place with background noise it is too soft. 8 - Have a model with a PSTN jack. There is a break out notch so that the phone can be used as a regular analog phone. Some H323 phones have this and it is very handy. 8 - Use better quality mouth pick transducers. The one used are too sensitive and clipping is noticeable. 9 - Mentioned before: The display is difficult to see, leave the back light on OR better still tilt the display up. My 2c contribution. MA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: Tuesday, 21 October 2003 2:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. 7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exten delay matching
Hi, After I hear the intro, I press 1 or 2 and I get a delay of about 5 seconds before the 1 or 2 exten is read. I am sure this worked without a delay before. I did a CVS upg about a week ago. I also just tried it with a single background statement, same result. Could be related to the DigitTimeout. Anyone having similar problems or is my logic warped. [from-pri] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,BackGround,intro-can exten = s,6,BackGround,intro-us exten = 1,1,agi,start-pri-can.agi exten = 2,1,agi,start-pri-us.agi exten = h,1,agi,end-pri.agi Thanks MA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: SIP i.e. Is something broken?
I filed a bug report yesterday about it. http://bugs.digium.com/bug_view_page.php?bug_id=330 Budgetones are effected, not sure about others. It seems to be codec related. If you use allow=all, then it tries to negotiate G723 with Ulaw and this effects other audio items. MA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Tuesday, 30 September 2003 9:20 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: SIP i.e. Is something broken? WipeOut wrote: Lists wrote: This is my issue as well, Does anyone know how to fix it? Roll back to the CVS from last Thurdsay, This worked for me.. If you like you could try Friday and see if it works which will help narrow down when the problem started.. :) I'm going to bet that it's codec negotiation. I posted a sip debug trace yesterday. I'm not inside the code to a degree that would let me nail anything down, but there were some things in there that lead me to think that asterisk doesn't think the Budgetone shares any codecs with it. . . It will be interesting to see what eventually transpires. B. -- This message has been scanned for viruses and is believed to be clean. Scan engine v4.2.40 for Linux. Virus data file v4294 created Sep 18 2003 Scanning for 80178 viruses, trojans and variants. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latest CVS breaks sound
Title: Message Hi, Checked out latest CVS and no sound from Playback, Background, MOHor bridged channels.mpg123 is active but no sound. Master
[Asterisk-Users] (no subject)
Hi, Checked out latest CVS and no sound from Playback, Background, MOH or bridged channels. mpg123 is active but no sound. Master ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP on TCP
Title: Message Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configuredand if so how. My service provider will only accept SIP signalling on TCP. Thanks Master
RE: [Asterisk-Users] SIP on TCP
JT, We use 2 providers iPCB.NET and NTT (backup) and both require signalling on TCP only. Interestingly, I find this to be the norm amongst Cisco powered providers. As * marches on to the #1 telco product and SIP to the #1 protocol of choice, protocol=[tcp,udp,auto] feature is a good idea in sip.conf. I will add it as a feature. Master -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, 4 September 2003 3:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP on TCP Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configured and if so how. My service provider will only accept SIP signalling on TCP. Thanks Master Out of curiosity, what SIP provider is that? I've never seen any SIP providers that even support SIP over TCP, much less mandate it. If that is required, maybe a protocol=[tcp,udp,auto] feature is a good idea in sip.conf. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
