[asterisk-users] WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Hi All,
I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to
this page
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ;
when I dial ,there have this warning:
-- Executing AsyncGoto("SIP/111-086497c8",
"SIP/113-08674628|dynamic-nway|111|1") in new stack
Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting
async goto (SIP/113-08674628) to dynamic-nway,111,1
Feb 2 16:53:10 DEBUG[4218]: channel.c:2834 ast_channel_masquerade: Planning to
masquerade channel SIP/113-08674628 into the structure of
AsyncGoto/SIP/113-08674628
Feb 2 16:53:10 DEBUG[4218]: channel.c:2847 ast_channel_masquerade: Done
planning to masquerade channel SIP/113-08674628 into the structure of
AsyncGoto/SIP/113-08674628
Feb 2 16:53:10 DEBUG[4218]: channel.c:2972 ast_do_masquerade: Got clone lock
for masquerade on 'SIP/113-08674628' at 0x8677314
Feb 2 16:53:10 DEBUG[4218]: channel.c:3154 ast_do_masquerade: Putting channel
SIP/113-08674628 in 2/2 formats
Feb 2 16:53:10 DEBUG[4218]: channel.c:3189 ast_do_masquerade: Released clone
lock on 'AsyncGoto/SIP/113-08674628'
Feb 2 16:53:10 DEBUG[4218]: channel.c:3198 ast_do_masquerade: Done
Masquerading SIP/113-08674628 (6)
Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge
failed on channels SIP/111-086497c8 and AsyncGoto/SIP/113-08674628
Feb 2 16:53:10 DEBUG[4218]: app_dial.c:1636 dial_exec_full: Exiting with
DIALSTATUS=ANSWER.
-- Executing Set("SIP/111-086497c8", "DYNAMIC_FEATURES=") in new stack
-- Executing Goto("SIP/111-086497c8", "dynamic-nway|111|1") in new stack
-- Goto (dynamic-nway,111,1)
== Channel 'SIP/111-086497c8' jumping out of macro 'nway-start'
Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge
failed on channels SIP/112-08641920 and SIP/111-086497c8
I want to know why there are this warning? How can I fix it?
With Regards,
Amy
李君
[EMAIL PROTECTED]
2007-02-02
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RE: [asterisk-users] Dell Servers
Dell 2950 doesn't come with Intel network chip. It comes with Broadcom extreme and module is BNX2. We are using these servers and using Gentoo linux. We are experiencing issues when there are 30+ calls enabled for call recording and we assume this is due to new SAS controller. If you not planning to use the servers for call recording then you should be ok. Ahsan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rousse Sent: 01 February 2007 19:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dell Servers Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --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] Dell Servers
It comes with PCI express by default, but you can ask dell to provide you the PCI raiser cards instead of PCI express. Ahsan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: 01 February 2007 20:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dell Servers Hello, I have installed Asterisk on several of them and there can be issues. Will you be installing a telco interface card on this server?(If so, which one) Will this server have PCI or PCIexpress expansion ports? MATT--- On 2/1/07, Eric Rousse <[EMAIL PROTECTED]> wrote: > Hi, > > I was planning on getting a Dell PowerEdge 2950 for our new Asterisk > configuration. > But while searching for documentation about it and/or reported issues, I > found this: > > http://www.voip-info.org/wiki/view/Asterisk+hardware > WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, > which has been known to cause random locksup - if you plan on using a > Dell server, disable the onboard controller and purchase an addon > ethernet card. > > Does anyone has real experience ? > > Thanks, > > -- > Eric Rousse > System Administrator > 514.380.2992 > 450.655.1001 > 1.888.641.5800 > > Telmatik inc. > 204 Montarville, suite 250 > Boucherville, QC, Canada > J4B 6S2 > > www.telmatik.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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --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] Dell Servers
> I bought a Dell 2850 as a pbx server and it just sucks IMHO > > The stupid thing has only 3 pci slots and even with only 3 > pci slots Dell > managed to have a shared irq on every slot, 1 for the scsi > controller and > one for each nic We're using a couple of Dell 1850's and I couldn't be happier with them. But then I don't use any Digium, Sangoma or other cards. We're running 100% VoIP through them. -- Andreas Sikkema ___ --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 Clone Asterisk
On 2/2/07, Robert DeVries <[EMAIL PROTECTED]> wrote: I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? Sometimes my installations goes to different directories. You should check first where are your files and what you make more (voicemail, monitor, etc) Conf: /etc/asterisk /etc/zaptel.conf Sounds: /usr/share/asterisk/sounds/ /var/lib/asterisk/sounds/ MOH: /usr/share/asterisk/mohmp3/ Logs: /var/log/asterisk/ AGIs: /var/lib/asterisk/agi-bin Database: /var/lib/postgresql -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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] Dell Servers
On Thu, 1 Feb 2007, Christophorus Laube wrote: We have a 2850 in a productive environment with a BNE1 performing well (OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu Edgy). You only have to blacklist some hotplug kernel modules and yes, we do have very long pings (1 ping per week with a check rate of 10min per SNMP). But that does happen very rare and I never noticed any dropped calls or bad audio quality. The 2850 is running on SCSI, the 2950 on an SAS RAID. In general I like the Dell machines, also with asterisk on them. The only thing is that Openmanage ist quite bad to install but that's nothing asterisk specific but linux related. Does that help? Would you be willing to share your blacklist for the kernel modules? 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] volume control in VoIP
Hi Is there a way to control volume in VoIP calls just like the "gain" parameters for ZAP lines? Thanks, François. ___ --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: CallerID to FXS (RE: [asterisk-users] SendText() question)
Eric "ManxPower" Wieling wrote: Leo Ann Boon wrote: Eric "ManxPower" Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. Tell that to my cheap analog caller id phone :) BTW, the sample zapata.conf in Asterisk also have the caller id names quoted. Maybe Mark can enlighten us :) Leo ___ --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 Clone Asterisk
On Friday 02 February 2007 12:29 am, Robert DeVries wrote: > Assuming I install Asterisk on the new machine, does anyone know what files > I would have to copy over? What comes to mind are the *.conf files in > /etc/asterisk, as well as the voicemail audio files. Anything else? Asterisk is actually pretty nice in the sense that all of the data is organized under very few directories: /etc/asterisk /usr/lib/asterisk /var/lib/asterisk /var/spool/asterisk if you want your logs: /var/log/asterisk If you want to copy the binaries you'll also need /usr/sbin/asterisk /usr/sbin/astgenkey /usr/sbin/aelparse /usr/sbin/autosupport /usr/sbin/muted /usr/sbin/rasterisk /usr/sbin/safe_asterisk /usr/sbin/smsq /usr/sbin/stereorize /usr/sbin/streamplayer And the man pages: /usr/share/man/man8/asterisk.8 /usr/share/man/man8/astgenkey.8 /usr/share/man/man8/autosupport.8 /usr/share/man/man8/safe_asterisk.8 Zaptel, however, isn't quite so pretty due to the nature of the beast: /etc/zapata.conf /sbin/ztcfg /lib/modules/`uname -r`/misc (the modules dir may have non-zaptel modules in there too, and you may have more tools than just ztcfg to copy over). If you're running a different kernel on the new machine, you'll have to rebuild the zaptel modules, naturally. Finally, if you have any custom scripts, sounds, callfile templates or other self-generated data outside of the directory paths mentioned above, you'll of course need to copy those, too. HTH, -A. ___ --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: CallerID to FXS (RE: [asterisk-users] SendText() question)
Yuan LIU wrote: From: Leo Ann Boon <[EMAIL PROTECTED]> Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in "Unknown"). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a "normal" line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid="John Doe" <1234> Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says "unknown" or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Can post your zapata.conf? You need to ensure Asterisk is sending the FSK signal at the right time. This is from my zapata.conf: signalling=fxo_ks sendcalleridafter=2 usecallerid=yes cidsignalling=bell cidstart=ring Leo ___ --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] Line drops
Hello to all, I post again (last time subject: Line drops strange problem(got event On hook) because i have caught in debug a situation where i get a call and the line drops and i get a call from the same caller and the line works well and the call normally closes by both parties. The only differences i find are underlined. If someone can understand the reason why the line drops from the debug messages or has any thoughts Any help would be highly appreciated . p.s. The whole situation is that my line drops (once or twice a week). If you want more details i can post them Thank you in advance *line drops situation --- * Jan 31 15:20:40 VERBOSE[25962] logger.c: -- SIP/51-0986fab0 is ringing Jan 31 15:20:40 DEBUG[25962] chan_zap.c: Requested indication 3 on channel Zap/7-1 Jan 31 15:20:40 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:40 DEBUG[25962] chan_zap.c: Got event Ring Begin(18) on channel 7 (index 0) Jan 31 15:20:42 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:42 DEBUG[25962] chan_zap.c: Got event Ring/Answered(2) on channel 7 (index 0) Jan 31 15:20:42 DEBUG[25962] chan_zap.c: Setting IDLE polarity due to ring. Old polarity was 0 Jan 31 15:20:45 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:45 DEBUG[25962] chan_zap.c: Got event Ring Begin(18) on channel 7 (index 0) Jan 31 15:20:46 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:46 DEBUG[25962] chan_zap.c: Got event Ring/Answered(2) on channel 7 (index 0) Jan 31 15:20:46 DEBUG[25962] chan_zap.c: Setting IDLE polarity due to ring. Old polarity was 0 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Setting NAT on RTP to 524288 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 52547: Match Found Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Setting NAT on RTP to 524288 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Checking SIP call limits for device 53 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: build_route: Contact hop: Jan 31 15:20:47 DEBUG[2434] channel.c: Avoiding initial deadlock for 'SIP/53-b7a05818' Jan 31 15:20:47 DEBUG[2442] channel.c: Planning to masquerade channel SIP/53-b7a05818 into the structure of SIP/51-0986fab0 Jan 31 15:20:47 DEBUG[2442] channel.c: Done planning to masquerade channel SIP/53-b7a05818 into the structure of SIP/51-0986fab0 Jan 31 15:20:47 DEBUG[25962] channel.c: Got clone lock for masquerade on 'SIP/53-b7a05818' at 0xb7a0adf4 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: update_call_counter(51) - decrement call limit counter Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Acked pending invite 102 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 31 15:20:47 DEBUG[25962] channel.c: Putting channel SIP/53-b7a05818 in 64/64 formats Jan 31 15:20:47 DEBUG[25962] channel.c: Released clone lock on 'SIP/51-0986fab0' Jan 31 15:20:47 DEBUG[25962] channel.c: Done Masquerading SIP/53-b7a05818 (0) Jan 31 15:20:47 VERBOSE[25962] logger.c: -- SIP/53-b7a05818 answered Zap/7-1 Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Requested indication -1 on channel Zap/7-1 Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Took Zap/7-1 off hook Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Enabled echo cancellation on channel 7 Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Engaged echo training on channel 7 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 52548: Match Found _Jan 31 15:22:40 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:22:40 DEBUG[25962] chan_zap.c: Got event On hook(1) on channel 7 (index 0) Jan 31 15:22:40 DEBUG[25962] chan_zap.c: disabled echo cancellation on channel 7 Jan 31 15:22:40 DEBUG[25962] channel.c: Didn't get a frame from channel: Zap/7-1 Jan 31 15:22:40 DEBUG[25962] channel.c: Bridge stops bridging channels Zap/7-1 and SIP/53-b7a05818 Jan 31 15:22:40 DEBUG[25962] chan_sip.c: update_call_counter(53) - decrement call limit counter Jan 31 15:22:40 DEBUG[25962] app_dial.c: Exiting with DIALSTATUS=ANSWER._ Jan 31 15:22:40 VERBOSE[25962] logger.c: == Spawn extension (ringoffice, s, 1) exited non-zero on 'Zap/7-1' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is '00381113237515' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is '00381113237515' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is 's' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is 'ringoffice' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is 'Zap/7-1' Jan 31 15:22:40 DEBUG[25962] pbx.c
[asterisk-users] install-misdn compile problem with debian
Hi, I get this error while compiling install-misdn on a Debian box with kernel 2.6.18: *make[2]: *** [/usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN/hfc_multi.o] Error 1* *make[1]: *** [_module_/usr/src/install-misdn/mISDN/drivers/isdn/hardware/mISDN] Error 2* *make[1]: Leaving directory `/usr/src/linux-headers-2.6.18-2-486'* *make: *** [MISDN_MAKE_MODS] Error 2 *anybody knows why? TIA Giorgio ___ --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 logging everything?
Hi, Is it possible to keep asterisk from logging exactly everything? I can do the logger rotate and keep the files small enough, but I think it's unneccesary to log exactly all data. File grows by about 5 gb per month! Thanks! Regards, Jan ___ --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 logging everything?
On 2/2/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Hi, Is it possible to keep asterisk from logging exactly everything? I can do the logger rotate and keep the files small enough, but I think it's unneccesary to log exactly all data. File grows by about 5 gb per month! Thanks! Regards, Jan ___ --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 Have a look at /etc/asterisk/logger.conf ___ --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 logging everything?
On Fri, Feb 02, 2007 at 12:29:34PM +0100, [EMAIL PROTECTED] wrote: > Hi, > > Is it possible to keep asterisk from logging exactly everything? I can > do the logger rotate and keep the files small enough, but I think it's > unneccesary to log exactly all data. > > File grows by about 5 gb per month! > Thanks! Don't use "safe_asterisk". If you do use it, set it to remove the 'v'-s from the default command-lines options of Asterisk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] windows SIP Softphones ?
Bruno Castelo Branco wrote: Hi Try that one http://www.counterpath.com/index.php?menu=Products&smenu=xlite Bruno C. Branco Do they still have the web-based one available, formerly "X-Web"? -- Chris Hills | Tel: +44 (0)1527 572754 IT Services | Fax: +44 (0)1527 572901 North East Worcestershire College | Web: http://www.ne-worcs.ac.uk/ smime.p7s Description: S/MIME Cryptographic Signature ___ --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] musiconhold restarts for every extension
Hi Brian! Actually i play a message to the caller, something like "Hello and welcome at ..., someone will take your call in a few seconds" - so a random musiclist is not an option. Even if i would play music only, it doesn't sound very cool if the music is changed within 5 secs. I would suggest to hang the DJ if i was the caller. Regards Christian On Thu, 1 Feb 2007 09:39:57 -0500 "Brian M. Arlinghaus" <[EMAIL PROTECTED]> wrote: > Benko, > > You can put multiple files in the MOH directory giving your listener > a good chance of getting a new piece of music each time he is on > hold. Asterisk picks one of your files randomly. > > Regards, > Brian > > - Original Message - > From: "Benko" <[EMAIL PROTECTED]> > To: > Sent: Wednesday, January 31, 2007 6:16 AM > Subject: Re: [asterisk-users] musiconhold restarts for every extension > > > > On Tue, 30 Jan 2007 12:04:30 -0600 > > "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> wrote: > >> > While in 1.2.9 musiconhold > >> > was playing continuous on sequential extensions after a > >> > timeout, it is restarted for every extension in 1.2.14: > >> > >> As I understand it, this is the way Native Music on Hold works. > >> mpg123 based MoH does not restart for each call. > > > > Well, it was working perfectly with Native MOH in 1.2.9. > > Judging the two replies, this is a bug(imho). I think it > > should at least be optional if you want it to be restarted or not(if > > there's anyone who needs the current behaviour). I don't want to > > sound like an dissatisfied customer however, i honour that asterisk > > is mostly voluntary work and since i'm not a programmer i try to > > contribute with feedback at least. > > > > Regards & Thx > > Christian > > ___ > > --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] How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms
*ping* I am interested in this too if anyone has any clues? I am looking to do this on a Cisco 7941/7961. Thanks, Steve On 1/26/07, Naija Man <[EMAIL PROTECTED]> wrote: Hello, We have an asterisk system with about 40 cisco 7940/7960 phones and a few linksys SPA941. I recently analyzed our network and discovered that the rtp packet size from the cisco phones is 10ms. We want to change the rtp packet size of the Cisco phones from 10ms to 20ms. I know how to do this on linksys phones and sipura ATAs but I cannot figure out how on the 7940/7960s. Is this possible? Does anyone have suggestions as to how I can do achieve this? Any tip or help will be appreciated. Codec: ULAW SIP firmware: 8.2 Thanks. Buki ___ --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] Re: Dell Servers
> "ER" == Eric Rousse <[EMAIL PROTECTED]> writes: ER> Hi, I was planning on getting a Dell PowerEdge 2950 for our new ER> Asterisk configuration. But while searching for documentation ER> about it and/or reported issues, I found this: ER> http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - ER> many Dell motherboards use the e1000 gigabit ethernet chipset, ER> which has been known to cause random locksup - if you plan on ER> using a Dell server, disable the onboard controller and purchase ER> an addon ethernet card. The e1000 ethernet chipset is basically the best you can get for Linux today. The lockup problem is more likely because Digium cards (and others) have issues with IRQ sharing. Maybe one day the telephony card vendors will wake up and realise that PCI IRQ sharing is perfectly well defined since PCI 1.0 and that the only problems with it are due to bugs in their drivers or their cards. /Benny ___ --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: CallerID to FXS (RE: [asterisk-users] SendText() question)
Leo Ann Boon wrote: Eric "ManxPower" Wieling wrote: Leo Ann Boon wrote: Eric "ManxPower" Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. Tell that to my cheap analog caller id phone :) BTW, the sample zapata.conf in Asterisk also have the caller id names quoted. Maybe Mark can enlighten us :) Since the telco never sends quotes on PSTN calls, I can't imagine how this could be the case. Remember, in most cases, quotes in Asterisk config files are considered part of the value. So if you did a callerid="Robert Dobbs" <5556661212> and then did a Noop($CALLERIDNAME) you would see ""Robert Dobbs"" on the Asterisk CLI. Maybe 1.2 and later silently strip off the quotes. ___ --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] volume control in VoIP
Yes. This is a function of the VoIP endpoint devices, not Asterisk. François Delawarde wrote: Hi Is there a way to control volume in VoIP calls just like the "gain" parameters for ZAP lines? Thanks, François. ___ --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] FreePBX/Debian Aborts Call While Connecting
You're looking at only the logfiles, which don't reflect the problem at
the other side, the switch which sees the incoming request abort before
it can complete the connection, and before the 45s timeout. What you're
missing is my reports of that difference on either side of the network,
which I have mentioned in every message to this list, including the one
you counted.
In any event, the problem is some kind of protocol handling bug, either
in the SIP server or the (SVN) version of Asterisk I'm using. I pointed
at a different (newer) SIP server at my same carrier, and have no
problem connecting. Though I was connecting OK to the old SIP server
with my old Asterisk version (1.2.12) before the "upgrade". I expect
that both the old SIP server and the SVN Asterisk version have bugs
which finally combine to abort improperly, and without proper failure
reporting by Asterisk.
Thanks anyway for trying to help.
On Thu, 2007-02-01 at 22:59 -0500, Asterisk wrote:
> On Thu, 2007-02-01 at 08:47 -0500, Matthew Rubenstein wrote:
> > The point is that the SIP carrier side gets the abort *before the SIP
> > carrier can complete the connection*. That doesn't take 45s. It takes
> > something like a few seconds. What is causing my (Asterisk) side to
> > abort right after completing registration?
> >
> >
> > On Thu, 2007-02-01 at 02:28 -0500, Asterisk wrote:
> > > Yeah, your waittime parameter in your call file is set to 45 seconds.
> > >
> > > db
> > >
> > > On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote:
> > > > I used the "FreePBX on Debian" HowTo at
> > > > http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
> > > > to initiate calls to my SIP carrier. They get my registration, but they
> > > > see that my call is interrupted before they can complete the connection.
> > > > My Asterisk log shows that the call times out after the time (45s)
> > > > specified in my dialplan Dial() command. What is wrong?
> > > >
> > > > [from /var/log/asterisk/full]:
> > [...]
>
> Alright, take a look the **Lines:
>
>
>
> **Line 1:
> Your dial sequence clearly shows the 45sec timeout value being applied
> as the second value in the dial plan "SIP/[EMAIL PROTECTED]|45| <<--
>
> Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing
> Dial("Local/[EMAIL PROTECTED],2", "SIP/[EMAIL PROTECTED]|45|
> M(say-call-2-digits^17182335097)g") in new stack
>
>
> **Line 2:
> The timer has expired 45000ms is the same 45 second timer that was set
> for timeout
>
> Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Nobody picked up in
> 45000 ms
>
> Line 3:
> The call is dropped towards the carrier.
>
>
> Maybe I am missing something here but it seems you are using a macro
> with some global variable set for a 45 second wait time for outbound
> calls.
>
>
> Thanks,
> Dave
>
--
(C) Matthew Rubenstein
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[asterisk-users] CallerID Name not available.
Hi,
An incoming call is redirected to another number by our asterisk
server. In the incoming call the caller name is present but when
redirect the call, the end receiver is not able to see the callerid
name. The caller id number is visible.
our related changes to extensions conf is below.
exten => {MY_EXT},14,Set(CALLERID(name)=OH ${CALLERID(name)})
exten => {MY_EXT},15,Dial(SIP/vitel-outbound/$[${MY_CONTACT}],30)
at 14 we try to present OH to the Caller Name. For eg "OH Shivram U".
(removing this line doesnt work too)at 14 we are able to see the
callerid name.
at 15 we route the call to our contact.
vitel-outbound is the service provider for the DIDs
MY_CONTACT would be the number we are trying to redirect to
The asterisk version we are using is 1.2.10. I tried passing the
option 'o' to the Dial command, but that doesnt work too.
Anything i'm missing out here.
Thanks in advance,
Shivram U
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Re: [asterisk-users] musiconhold restarts for every extension
On Tue, 30 Jan 2007 17:50:53 +0100 Benko <[EMAIL PROTECTED]> wrote: > Hello! > > I've upgraded from 1.2.9 to 1.2.14 recently but experience an > unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was > playing continuous on sequential extensions after a > timeout, it is restarted for every extension in 1.2.14: > > ;music starts > exten => 902,1,Dial(SIP/[EMAIL PROTECTED]|5|m(mymusic)) > ;music starts again > exten => 902,n,Dial(SIP/[EMAIL PROTECTED]|5|m(mymusic)) > ;and again > exten => 902,n,Dial(SIP/[EMAIL PROTECTED]|5|m(mymusic)) > > In the changelog this is not mentioned, also the bugs related to > changes in musiconhold.c don't seem to have anything to do with my > issue... > Is there a setting i didn't see which changes this behaviour? It is > quite annoying for the caller and unprofessional if he hears the same > moh-file again after some seconds, imho this is not a correct > behaviour... > > I recognized that this also happens in 1.4, which i'm currently only > testing. > > thx > christian I've filed a bugreport in the meanwhile, hope there'll be a resolution -> http://bugs.digium.com/view.php?id=8969 regards Benko ___ --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] FOP (or equivalent) and timers
Hi, Is it easy to show calls elapsed duration within a FOP button ? FOP documentation mentions "timer" but I couldn't find any example or clue proving it's possible to do what I'm looking for. Anyway, would you recommend another software to customize Asterisk call display ? Regards ___ --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] No RTP packets received by Asterisk when calling SIP to SIP
I have the following setup: UA1 (SPA2000) -- Nat1 -- Asterisk (public internet) -- Nat 1 -- UA2 (X-Lite) Relevant parts of sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip = 60.234.100.100 ;External IP address localnet = 192.168.1.0/255.255.255.0;Local network address allow=all [1590] username=1590 type=friend secret=secret qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=test canreinvite=no allow=all [1593] username=1593 type=friend secret=secret qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=test canreinvite=no allow=all I have enabled rtp debugging and notice that Asterisk is receiving no rtp traffic. When I call from either UA to voicemail for example I see RTP traffic e.g. call from 1590 Got RTP packet from 60.234.200.200:38510 (type 0, seq 1245, ts 207620, len 160) Sent RTP packet to 60.234.200.200:38510 (type 0, seq 61963, ts 34880, len 160) e.g. call from 1593 Got RTP packet from 60.234.200.200:16470 (type 0, seq 892, ts 316685167, len 240) Sent RTP packet to 60.234.200.200:16470 (type 0, seq 1156, ts 15360, len 160) I thought that with canreinvite=no all audio would go through Asterisk. What have I missed? Asterisk 1.2.13 Fedora Core 5 Regards Cameron ___ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html___ --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] Local channel with /n doesn't hangup after transfer. Why?
Hello all
I asked similar question some time ago but didn't get answer... Maybe
this should asked in asterisk-dev or bugs.digium.com?
For example, I have 3 sip phones defined in sip.conf - 101, 103, 109 and
this simple dialplan:
[local-ext]
exten => 101,1,Dial(SIP/101,,t)
exten => 107,1,Dial(SIP/107,,t)
exten => 109,1,Dial(SIP/109,,t)
exten => 1109,1,Dial(Local/[EMAIL PROTECTED]/n,,t)
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/101-0935e788",
"Local/[EMAIL PROTECTED]/n||t") in new stack
-- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2",
"SIP/109||t") in new stack
SIP/101 dials 1109. SIP/109 answers the call. We have 4 active channels:
Channel Location State Application(Data)
SIP/109-0935cfd0 (None) Up Bridged
Call(Local/[EMAIL PROTECTED]
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up Dial(SIP/109||t)
Local/[EMAIL PROTECTED] (None) Up Bridged
Call(SIP/101-0935e788)
SIP/101-0935e788 [EMAIL PROTECTED]:1 Up
Dial(Local/[EMAIL PROTECTED]/n||t)
Now, SIP/109 transfers call to SIP/107. SIP/107 ansfer and we still have
4 channels:
Channel Location State Application(Data)
SIP/107-093931f0 (None) Up Bridged
Call(Local/[EMAIL PROTECTED]
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up Dial(SIP/107||t)
Local/[EMAIL PROTECTED] (None) Up Bridged
Call(SIP/101-0935e788)
SIP/101-0935e788 [EMAIL PROTECTED]:1 Up
Dial(Local/[EMAIL PROTECTED]/n||t)
Channel Local/[EMAIL PROTECTED] is quite a problem when we use Local channel
in queues, because queue thinks that's it's busy even after transfer.
Why is this channel doesn't hangup and SIP/101 and SIP/107 can't be
bridged without it?
AgentCallbackLogin is depreciated and it's not possible to make normal
callback agents with such work of Local channel.
If we don't use /n then queue knows nothing about the state of agent.
That's not very good too.
Thank you.
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[asterisk-users] RE: [SOLVED] Dial option G - Passing parameters?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Collins
> Sent: Thursday, February 01, 2007 12:38 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Dial option G - Passing parameters?
>
> Has anyone used the G option with the Dial app? I'm looking for a way
> to control the called party leg. Specifically, I'd like to pass a few
> variables to the called side for some call control. Here's a synopsis
> of what I'm doing:
>
> Make outbound call w/ AMI Originate action.
> Called party answers ("Customer")
> Customer identifies himself, and now I use Dial w/ the G option:
> Dial(Zap/g9/${agentext}|60|mG(Agent_Xfer^s^1)
> Customer hears MOH while the Dial app gets the "agent" on the line
>
> My destination context looks like this:
> [Agent_Xfer]
> exten => s,1(Customer),Meetme({$ConfRoom}|qM)
> exten => s,2(Agent),Macro(Connect_to_agent,${Customerid},${ConfRoom})
>
> Customerid and ConfRoom are channel variables that are set in the
> Originate action and at the start of the dialplan processing,
> respectively.
>
> The idea is to put the customer in a conference room, listening to
MOH,
> until I can get an agent on the line. (This part works pretty well.)
> The agent is an extension on a legacy PBX, so a simple Dial with a
macro
> has undesired side effects. (Specifically, the customer hears ringing
> or the legacy PBX's MOH, depending upon the status of the transfer.)
> Putting the customer in a conf room, listening to music, is the best
> solution I can think of.
>
> The problem is that I don't know how to get the two channel variables
> over to the "Agent" leg of the call. I don't see anything in the docs
> about the G option accepting arguments to pass to the called leg. Is
> there any way that I can get the two variables' values over to the
> called leg?
>
> -MC
FYI,
After some researching I realized that I did not understand variable
inheritance. I've 'globalized' the two variables in question so that
they are inherited by the second leg of the call. For your reference,
the helpful information was found on the wiki:
http://www.voip-info.org/wiki-Asterisk+variables
(Specifically under the heading "Inheritance of Channels Variables."
-MC
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[asterisk-users] queues and LOCAL for members
Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. best regards Thomas ___ --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] Re: unable to create channel, in strange state, exited non-zero, etc.
On 1/25/07, Wayne Jensen <[EMAIL PROTECTED]> wrote: I'm having various issues that may or may not be related to each other (I'm pretty sure they are). We've had this system for a year now (quad T1 card, right now we have 1 T1 coming in, 2 going out to channel banks) and we've had intermittent "ghost" calls--it appears that what is happening is a call is made, the number being called is disconnected/busy/answering machine/whatever, we hang up and the phone starts ringing. Answer the phone and it's that same call still not hung up. That problem hadn't happened very often so we didn't worry too much about it. It was just a little annoying. We had pretty low traffic on the system though and were mostly still using our old phone system. On Monday I switched so that all of the phones are going through this Asterisk system and Tuesday morning we started having major problems. Calls were being dropped in the middle of the call. At times, everyone who was trying to make a call would get a "fast busy" and "Unable to create channel of type 'Zap'" showed up in the logs. Yesterday we had various Red and Yellow alarms. Here are a few lines from the logs: Yesterday: Jan 24 10:01:47 WARNING[12435] chan_zap.c: Detected alarm on channel 24: Red Alarm Jan 24 10:02:11 NOTICE[25813] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Today: Jan 25 08:39:23 WARNING[6863] chan_zap.c: zt hook failed: Device or resource busy Jan 25 08:40:32 WARNING[932] chan_zap.c: Ring/Off-hook in strange state 6 on channel 2 From console: == Spawn extension (phones-agent, 1510272, 2) exited non-zero on 'Zap/95-1' Still having problems here. Today there was a red alarm on channels 1-24 (the T1 going out to the telco) and all calls were dropped. This has also happened several other times. We've been using this T1 without the asterisk box for over a year now and this has never happened before, and even now if I disconnect the asterisk box and run the line straight into a channel bank it doesn't happen. I've googled and googled and can't figure it out. E&M Wink signalling B8ZS/ESF Thanks! Wayne ___ --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] volume control in VoIP
Don't you think it could be an interesting feature in Asterisk? It already does transcoding, why not gain when voice flow passes through it? François. Eric "ManxPower" Wieling wrote: Yes. This is a function of the VoIP endpoint devices, not Asterisk. François Delawarde wrote: Hi Is there a way to control volume in VoIP calls just like the "gain" parameters for ZAP lines? Thanks, François. ___ --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
[asterisk-users] 1.4 res_snmp dependencies (Debian)
I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box running Debian Sarge. res_snmp says its dependencies are netsnmp but Debian doesn't seem to have a netsnmp package. I've tried installing pretty much every package available related to snmp and no luck. I'm just wondering if anyone has successfully built the res_snmp module under Debian Sarge stable. Any help or suggestions are appreciated. ___ --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] volume control in VoIP
Perhaps you can write the functionality? I'm sure you can do a quick hack of you modify app_voicechangedial. On 2/2/07, François Delawarde <[EMAIL PROTECTED]> wrote: Don't you think it could be an interesting feature in Asterisk? It already does transcoding, why not gain when voice flow passes through it? François. Eric "ManxPower" Wieling wrote: > Yes. This is a function of the VoIP endpoint devices, not Asterisk. > > François Delawarde wrote: >> Hi >> Is there a way to control volume in VoIP calls just like the "gain" >> parameters for ZAP lines? >> Thanks, >> François. >> ___ >> --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 ___ --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] API Originate Action - distinguishing betweenNoAnswer and Invalid phone number
I have been having a very similar problem. Has anyone here gotten a
DIALSTATUS for calls started with originate?
I did some research and saw some posts that local channels are the solution
to this problem. However, I could not find examples of how to use local
channels with originate. I could not get it to work. I posted a topic (Using
Local Channels with originate) to this list yesterday with the details about
what I had tried. Maybe you will see what I missed.
-Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Collins
Sent: Thursday, February 01, 2007 11:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] API Originate Action - distinguishing
betweenNoAnswer and Invalid phone number
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork
Sent: Thursday, February 01, 2007 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] API Originate Action - distinguishing between
NoAnswer and Invalid phone number
On 2/1/07, Michael Collins <[EMAIL PROTECTED]> wrote:
Is there a way to distinguish between a no answer and an invalid? For
me, a 'failed' attempt is dialing an invalid number, and I'd like the
CDRs to reflect that. I'd like a no answer to be as 'successful' as a
busy.
The ${DIALSTATUS} channel variable stores the result of the dial attempt:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
You can store it on the CDR's userfield column using the cdr function:
Set(CDR(userfield)=${DIALSTATUS})
Actually, I can't. The dialplan execution goes straight to the 'failed'
extension. When it does so, the DIALSTATUS variable gets cleared out. I
have this in my dialplan:
exten => failed,n,Noop(Dial status is '${DIALSTATUS}')
The log yields this:
-- Executing NoOp("OutgoingSpoolFailed", "Dial status is ") in new stack
Is there perhaps a way to make DIALSTATUS persist or get populated when the
dialplan hits the failed extension?
-MC
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Re: [asterisk-users] Talkoff
McGhee, Stefano wrote: > Hello all, Please don't reply to an existing message to start a new topic. It screws up the message threading. Thanks, -Stephen- ___ --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] Zap Load/Stress scripts?
Porier, Jeremy M. wrote: > Are there any scripts out there that would help me stress test two boxes > that are setup back to back with 4 PRI connections? We're having > problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm > tired of "testing" them in a production environment. As Sangoma > provides firmware updates (and various other shots in the dark) I'd like > to be able see if the problem is fixed in an isolated environment. I > just need a way to simulate call volume on 4 t1s. Please don't reply to an existing message to start a new topic. It screws up the message threading. Thanks, -Stephen- ___ --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] 1.4 res_snmp dependencies (Debian)
On Fri, Feb 02, 2007 at 12:49:26PM -0600, Jeremiah Millay wrote: > I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box > running Debian Sarge. res_snmp says its dependencies are netsnmp but > Debian doesn't seem to have a netsnmp package. I've tried installing > pretty much every package available related to snmp and no luck. I'm > just wondering if anyone has successfully built the res_snmp module > under Debian Sarge stable. Any help or suggestions are appreciated. In Sarge: libsnmp5-dev In Etch: libsnmp9-dev In any case, 'apt-get install libsnmp-dev' would work. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] 3 PCI slot with exclusive IRQ ? please advice!
Leo Ann Boon wrote: > Alessio Focardi wrote: >> Hi, >> >> I'm looking for an hardware platform for an * installation that should >> have at least 3 PCI slot with no irq sharing whatsoever. >> > Use an industrial PC with a backplane bus. You can easily get 3-4 usable > slots in a 2U and 10-14 slots if you use a 4U... ...and have zillions of dollars :) Industrial PCs are pretty expensive. -Stephen- ___ --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] kewlstart disconnect threshold
Hi: Tzafrir Cohen wrote: > On Wed, Jan 31, 2007 at 07:57:54PM -0700, Stephen Bosch wrote: >> Hi, folks: >> >> Can the loop drop detection threshold (normally defined in milliseconds) >> be set on the Digium TDM-400 cards? Most PBXs let you set this value. > > What exactly do you need it for? > > On the FXO module (detecting) or on the FXS module (generating)? On the FXO module. The loop drop comes from the switch, after a remote party hangup. This is part of Calling Party Control (CPC). Leo Ann Boon wrote: > Stephen Bosch wrote: >> Hi, folks: >> >> Can the loop drop detection threshold (normally defined in milliseconds) >> be set on the Digium TDM-400 cards? Most PBXs let you set this value. >> > Good question. Anyone knows if the TDM-400 actually detect loop drops? Well, that's really what kewlstart (and loopstart) means. If it couldn't, then Asterisk wouldn't know that the call had been hung up, and hog the channel. The question is whether the detection threshold (how long the loop has to drop before the Zap driver detects the hangup condition) can be adjusted somewhere. Perhaps it's in the driver source? -Stephen- ___ --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 server RFC conformance
Hi Asterisk Gurus, I am new to Asterisk server. We are trying to use Asterisk for testing one of our new products. I was wondering if anyone can tell me if it is RFC compliant or how can i use Asterisk to test it for some basic RFC compliance. Thanks in Advance, ___ --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] Zap Load/Stress scripts?
hi, did you think to tx & rx params you can consult asterisktutorials.com, i m not sure of this but may be it will work 2007/2/2, Stephen Bosch <[EMAIL PROTECTED]>: Porier, Jeremy M. wrote: > Are there any scripts out there that would help me stress test two boxes > that are setup back to back with 4 PRI connections? We're having > problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm > tired of "testing" them in a production environment. As Sangoma > provides firmware updates (and various other shots in the dark) I'd like > to be able see if the problem is fixed in an isolated environment. I > just need a way to simulate call volume on 4 t1s. Please don't reply to an existing message to start a new topic. It screws up the message threading. Thanks, -Stephen- ___ --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] 3 PCI slot with exclusive IRQ ? please advice!
no you can look for tyan machines they aren't expencive 2007/2/2, Stephen Bosch <[EMAIL PROTECTED]>: Leo Ann Boon wrote: > Alessio Focardi wrote: >> Hi, >> >> I'm looking for an hardware platform for an * installation that should >> have at least 3 PCI slot with no irq sharing whatsoever. >> > Use an industrial PC with a backplane bus. You can easily get 3-4 usable > slots in a 2U and 10-14 slots if you use a 4U... ...and have zillions of dollars :) Industrial PCs are pretty expensive. -Stephen- ___ --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] asterisk server RFC conformance
A S wrote: > Hi Asterisk Gurus, > I am new to Asterisk server. We are trying to use Asterisk for testing > one of our new products. I was wondering if anyone can tell me if it is > RFC compliant or how can i use Asterisk to test it for some basic RFC > compliance. > Thanks in Advance, That's an awfully broad question. Which RFCs are you thinking of in particular? -Stephen- ___ --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] Problem with Voipjet ...
Voipjet locks $1.2 per running call and unlocks when call ends .. so $12 = 10 simultaneous calls ( if rate is 1.2 cents ) . On 02/02/07, Robert DeVries <[EMAIL PROTECTED]> wrote: I have found that if you don't have the minimum balance required for the voipjet "premium" server, you get the "circuits busy" message, you might want to check your balance. On 1/30/07, Alejandro Lengua <[EMAIL PROTECTED]> wrote: > > Hello, we have this problem with Trixbox 1.23 > I have created an outgoing route where the 1st line > has Voipjet and the 2nd an 3rd have voipcheap accounts. > > The problem is that at certain moments, when we call all > the calls go through the voipcheap SIP accounts SIP, whose > quality are not only not good enough but also consume a lot > of bandwidth. > > The error message that returns Voipjet to Asterisk is > that all "circuits busy". What I asume from this? > > Thanks in advance > ___ > --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] Zap Load/Stress scripts?
On 1 Feb 2007, at 16:34, Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two
boxes
that are setup back to back with 4 PRI connections? We're having
problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm
tired of "testing" them in a production environment. As Sangoma
provides firmware updates (and various other shots in the dark) I'd
like
to be able see if the problem is fixed in an isolated environment. I
just need a way to simulate call volume on 4 t1s.
Olle had a script for testing SIP which involved bouncing a call
between the
dialplans of 2 systems 'one' and 'two', incrementing the extension
number.
You could use a similar thing for ZAP, something like:
;extensions.conf on one
[bounce]
exten => _[0-5][0-9],1,dial(zap/g1/{$EXTEN})
;extensions.conf on two
[increment]
exten => _[0-5][0-9],1,dial(zap/G1/{$EXTEN}+1)
exten => 60,dial(sip/SecondPhone)
So you place a call to one, extension 00 from a sip phone,
it calls two on extension 00, which calls 01 on one etc untill you have
all 120 channels tied up, finally placing the call to your second SIP
phone.
The great thing about this is :
1) you get to hear the audio quality of 120 calls at the same time.
2) the call setup/teardown all happens at once, stress testing the
link pretty
effectively.
(Warning - beer has passed my lips during the creation of this email
- trust at your peril)
Tim.
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[asterisk-users] 7912 issues half audio
I have a 6 - 7912's. I have a TDM2402E echo cancel card. Asterisk 1.2.12.1 Some times extension to extension the audio is only heard one way. Some times extension to TDM2402 calls audio is only heard one way. I have turned off the echo suppression on the 7912's config. Any idea why it would be doing this? Something to try? THanks, Jerry ___ --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] API Originate Action - distinguishingbetweenNoAnswer and Invalid phone number
> I have been having a very similar problem. Has anyone here gotten a > DIALSTATUS for calls started with originate? > > I did some research and saw some posts that local channels are the > solution > to this problem. However, I could not find examples of how to use local > channels with originate. I could not get it to work. I posted a topic > (Using > Local Channels with originate) to this list yesterday with the details > about > what I had tried. Maybe you will see what I missed. > > -Brian > Brian, I have had zero success with local channels as well. When I dial a local channel, I actually get TWO outbound channels. It's weird. My logs show two passes through the dialplan even though I've called Dial(Local/xxx) only once. The phone number received two calls simultaneous. I've tried with and without the /n just to see if there's a difference. (There isn't, at least on my system.) If anyone out there has success stories using local channels with API Originate (or .call files) then we'd love to hear about it! Please let us know how you've overcome the limitations of autodialing, i.e. no DIALSTATUS, no dialplan processing on failed attempts unless you have a 'failed' extension, no DIALSTATUS information going to the 'failed' extension, etc. I still don't know how to distinguish between a legit NO ANSWER and an INVALID phone number. (They both 'fail' on an Originate and they both produce a second CDR with a disposition of NO ANSWER. BTW, I'm using PRI, and I've tried both inband and outofband signaling.) Thanks, MC ___ --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] musiconhold restarts for every extension
On Fri, 2 Feb 2007 16:56:59 +0100 Benko <[EMAIL PROTECTED]> wrote: > > I've filed a bugreport in the meanwhile, hope there'll be a resolution > -> http://bugs.digium.com/view.php?id=8969 issue was resolved in 1.2 rev 53084 and 1.4 rev 53088 (see http://bugs.digium.com/view.php?id=8672) but there's another issue in 1.4 rev. 53114, see my bugreport quoted above for details... regards benko ___ --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] queues and LOCAL for members
On 2/2/07, Thomas Winter <[EMAIL PROTECTED]> wrote: Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. There's been some efforts to have Local channels as viable queue members. I'm not quite sure that I understand your issue. Can you post some more details possibly in a bug on bugs.digium.com ? Thanks. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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
RE: [asterisk-users] musiconhold restarts for every extension
The problem can be reproduced in the same way by putting a caller on hold, unholding, and holding again. The MOH restarts from the beginning of whichever file it was playing last. (I have random enabled, so it randomly picks a "please wait for the next blah blah blah" file). (I'm using 1.4 release). Does this occur for you as well? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Benko > Sent: Friday, February 02, 2007 5:02 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] musiconhold restarts for every extension > > On Fri, 2 Feb 2007 16:56:59 +0100 > Benko <[EMAIL PROTECTED]> wrote: > > > > I've filed a bugreport in the meanwhile, hope there'll be a resolution > > -> http://bugs.digium.com/view.php?id=8969 > > > issue was resolved in 1.2 rev 53084 and 1.4 rev 53088 (see > http://bugs.digium.com/view.php?id=8672) but there's another issue in > 1.4 rev. 53114, see my bugreport quoted above for details... > > regards > benko > ___ > --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] problems with SJPhone (I feel stupid about this)
have a Grandstream and SJPhone SIP phones going to asterisk. with SJPhone (on Linux) getting. any ideas?? SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKc0a80264001045c3c2c52331d4920678;received=24.10.123.39;rport=60754 From: ;tag=22261807771886928353 To: ;tag=as45966c6b Call-ID: [EMAIL PROTECTED] CSeq: 41 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 - Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.___ --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] 3 PCI slot with exclusive IRQ ? please advice!
Stephen Bosch wrote: ...and have zillions of dollars :) Industrial PCs are pretty expensive. Over here, they're actually quite reasonably priced. A 2U rackmount P4 D930 3.0GHz, 1GB RAM system with 4 PCI (32bit) slots starts around US$1K. Leo ___ --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] kewlstart disconnect threshold
Good question. Anyone knows if the TDM-400 actually detect loop drops? Well, that's really what kewlstart (and loopstart) means. If it couldn't, then Asterisk wouldn't know that the call had been hung up, and hog the channel. For loopstart lines, I don't think Asterisk detects loop drops. If it does, we won't have lots of people complaining about asterisk not hanging up when the remote party hangs up. a quick grep of the asterisk source turns up only chan_vpb has any mention of loop drop, not in chan_zap nor in the zaptel driver. Leo ___ --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] CallerID Name not available.
Shivram u wrote: Hi, An incoming call is redirected to another number by our asterisk server. In the incoming call the caller name is present but when redirect the call, the end receiver is not able to see the callerid name. The caller id number is visible. If you're calling PSTN, caller id name is not guaranteed to be supported. Leo ___ --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] Call Waiting broken on ZAP
Problem: *Call* *waiting* comes in, I press flash to answer it, and the
first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP.
System:
Analog stations and trunks running on a pair of TDM400's. It does NOT have *
call* *waiting* from the phone company, and I have enabled it in all my conf
files. The trunks are set to FXSKS and the stations are FXOKS. I am not
using *call* progress or busy detect. Also its * 1.2.13 w/ FreePBX2.2. I
have scoured the net for this, and nobody seems to know.
Here is some logging from a *call*:
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Requested indication 3 on channel
Zap/5-1
Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Function result is 's'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Function result is '5'
Feb 1 17:41:53 DEBUG[6765] db.c: Unable to find key '5187152626' in family
'blacklist'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Took Zap/5-1 off hook
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Enabled echo cancellation on channel
5
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Engaged echo training on channel 5
Feb 1 17:41:54 DEBUG[6765] channel.c: Scheduling timer at 160 sample
intervals
Feb 1 17:42:00 DEBUG[6765] chan_zap.c: DTMF digit: 5 on Zap/5-1
Feb 1 17:42:00 DEBUG[6765] channel.c: Scheduling timer at 0 sample intervals
Feb 1 17:42:00 DEBUG[6765] pbx.c: Oooh, got something to jump out with
('5')!
Feb 1 17:42:01 DEBUG[6765] chan_zap.c: DTMF digit: 0 on Zap/5-1
Feb 1 17:42:02 DEBUG[6765] chan_zap.c: DTMF digit: 0 on Zap/5-1
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '5187152626'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '5187152626'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '5187152626/user' in
family 'DEVICE'
Feb 1 17:42:02 DEBUG[6765] func_db.c: DB: DEVICE/5187152626/user not found
in database.
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '/cidname' in family
'AMPUSER'
Feb 1 17:42:02 DEBUG[6765] func_db.c: DB: AMPUSER//cidname not found in
database.
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '-1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '64'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '"" <5187152626>'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CFU'
Feb 1 17:42:02 DEBUG[6765] func_db.c: DB: CFU/500 not found in database.
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '15'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 VERBOSE[6765] logger.c:
recordingcheck|20070201-174202|1170380512.151: Inbound recording enabled.
Feb 1 17:42:02 VERBOSE[6765] logger.c:
recordingcheck|20070201-174202|1170380512.151: CALLFILENAME=
20070201-174202-1170380512.151
Feb 1 17:42:02 DEBUG[6765] channel.c: Spy MixMonitor added to channel
Zap/5-1
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: Starting New
Dialparties.agi
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: priority is 1
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: Caller ID name is
'unknown' number is '5187152626'
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: Methodology of ring
is 'none'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CF'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'DND'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CFB'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CFU'
Feb 1 17:42:02 DEBUG[6765] chan_zap.c: Requested indication 3 on channel
Zap/5-1
Feb 1 17:42:02 DEBUG[6765] channel.c: Building translator from ulaw to
SLINEAR for spies on channel Zap/5-1
Feb 1 17:42:03 DEBUG[6765] chan_zap.c: Exception on 13, channel 2
Feb 1 17:42:03 DEBUG[6765] chan_zap.c: Got event Ring/Answered(2) on channel
2 (index 0)
Feb 1 17:42:03 DEBUG[6765] chan_zap.c: Enabled echo cancellation on channel
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
From: Leo Ann Boon <[EMAIL PROTECTED]> Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid="John Doe" <1234> Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says "unknown" or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Can post your zapata.conf? Forgot to mention, I was referring to calling from Asterisk itself (like using console) to another Asterisk via SIP without registration. (Same as the original post.) Unlike a soft or hard SIP phone, Asterisk's sip.conf has only one parameter callerid. Yuan Liu You need to ensure Asterisk is sending the FSK signal at the right time. This is from my zapata.conf: ... Leo ___ --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] volume control in VoIP
From: "Andrew Joakimsen" <[EMAIL PROTECTED]> Perhaps you can write the functionality? I'm sure you can do a quick hack of you modify app_voicechangedial. Not sure if this is a good idea. How do you handle situations where no transcoding is required? You don't want unnecessary heavy lifting. Yuan Liu On 2/2/07, François Delawarde <[EMAIL PROTECTED]> wrote: Don't you think it could be an interesting feature in Asterisk? It already does transcoding, why not gain when voice flow passes through it? François. Eric "ManxPower" Wieling wrote: > Yes. This is a function of the VoIP endpoint devices, not Asterisk. > > François Delawarde wrote: >> Hi >> Is there a way to control volume in VoIP calls just like the "gain" >> parameters for ZAP lines? >> Thanks, >> François. >> ___ >> --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 ___ --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
