Re: [asterisk-users] outging ---asterisk -bug
jordan pan schrieb: Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid: 15015895665 [myivr] exten = s,1,Background(test) exten = s,n,WaitExten Thanks in advance! -- Best regards! jordan pan Location:Shenzhen China Company:www.justcall.cn http://www.justcall.cn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, you should have a look at originate CLI command. you can call using asterisk -rx originate Zap/g0/1501589... Background(test). and maybe you should try an answer before starting the background in myivr. best regards. steve smith -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // s...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
2008/12/23 Yehavi Bourvine yehavi.bourv...@gmail.com I have one ST2030 bought for testing. Indeed it has a very intuitive user's interface, bue I've found two drawbacks: - Its sound quality has some place to be improved... - It has no RPID support (displaying the name of the called party). Hi, Have you tried P-Asserted-Identity features (from latest Asterisk and ST2030) ? I 'm hoping that using this feature, you can have your ST2030 displaying the name of the person you're calling : whenever, Asterisk is acking the INVITE message it received from the ST2030, it should include the name of called party. To be honest, I really don't know how you can teach Asterisk to behave this way (ie to include called party name in ACK replies) but I'm sure the phone support displaying names from trusted parties. It's on my TODO list, anyway, as I would also like to let this phone display caller's name when picking up a call (instead of the dialed string). Regards - If these two issues are fixed, then it might be the better choice for cheaper price. __Yehavi: 2008/12/21 Olivier oza-4...@myamail.com I don't know if Thomson ST2030 SIP phones are distributed where you live but those have the best feature set-price ratio. They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...) with up to 5 simultaneous calls. Here in France, those are selected everywhere ... I would recommend them without any hesitation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
I'll have to run some TCPDUMP to see what happens. I'll also try this with OpenSIPS where there is more flexbility with the header fields. Thanks, __Yehavi: 2008/12/23 Olivier oza-4...@myamail.com 2008/12/23 Yehavi Bourvine yehavi.bourv...@gmail.com I have one ST2030 bought for testing. Indeed it has a very intuitive user's interface, bue I've found two drawbacks: - Its sound quality has some place to be improved... - It has no RPID support (displaying the name of the called party). Hi, Have you tried P-Asserted-Identity features (from latest Asterisk and ST2030) ? I 'm hoping that using this feature, you can have your ST2030 displaying the name of the person you're calling : whenever, Asterisk is acking the INVITE message it received from the ST2030, it should include the name of called party. To be honest, I really don't know how you can teach Asterisk to behave this way (ie to include called party name in ACK replies) but I'm sure the phone support displaying names from trusted parties. It's on my TODO list, anyway, as I would also like to let this phone display caller's name when picking up a call (instead of the dialed string). Regards - If these two issues are fixed, then it might be the better choice for cheaper price. __Yehavi: 2008/12/21 Olivier oza-4...@myamail.com I don't know if Thomson ST2030 SIP phones are distributed where you live but those have the best feature set-price ratio. They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...) with up to 5 simultaneous calls. Here in France, those are selected everywhere ... I would recommend them without any hesitation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Tue, Dec 23, 2008 at 1:48 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: I have one ST2030 bought for testing. Indeed it has a very intuitive user's interface, bue I've found two drawbacks: Its sound quality has some place to be improved... It has no RPID support (displaying the name of the called party). If these two issues are fixed, then it might be the better choice for cheaper price. __Yehavi: 2008/12/21 Olivier oza-4...@myamail.com I don't know if Thomson ST2030 SIP phones are distributed where you live but those have the best feature set-price ratio. They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...) with up to 5 simultaneous calls. Here in France, those are selected everywhere ... I would recommend them without any hesitation. ___ While feeling cheap, the BT101 seemed reasonable during testinng. It was the day to day customer issues that made me go back and replace with Polycom for hardware cost only, no labor. I guess you have to define testing. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why does users.conf generate SIP peer and SIP user?
Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?
It's all ball bearings these days On Tue, Dec 23, 2008 at 4:35 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
On Mon, Dec 22, 2008 at 09:33:18AM -, Andrew Thomas wrote: You don't really need to use any local MTA if you use the sendEmail script. I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/ Which is essentially the same as using ssmtp / esmtp / nullmailer (non-queuing sendmail replacements). The mentioned prorams actually give you the same command-line interface as sendmail, and hence require less (if at all) change of voicemail.conf. This actually works by 'talking' directly to any SMTP server - even remote ones (I use our Exchange server for our e-mails). And this is a reminder: they don't queue mail. Hence if they fail to deliver once, the mail is lost. May not be the best idea for sending mail over the internet. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4with Lenny
On Mon, Dec 22, 2008 at 10:46:46AM +0100, Olivier wrote: Hi Andrew, 2008/12/22 Andrew Thomas a...@datavox.co.uk JFYI - I run (successfully) agx-addons with 1.4.22 and Etch. Make sure you have the right version of SpanDSP installed (as well as the tiff libraries). which are (thinking of both SpanDSP and libiff) ? libtiff, I guess. $ rmadison libspandsp-dev libspandsp-dev | 0.0.2pre26-1 | etch-m68k | m68k libspandsp-dev | 0.0.2pre26-1 |stable | alpha, amd64, arm, hppa, i386, ia64, mips, mipsel, powerpc, s390, sparc libspandsp-dev | 0.0.5~pre4-1 | testing | alpha, amd64, arm, armel, hppa, i386, ia64, mips, mipsel, powerpc, s390, sparc libspandsp-dev | 0.0.5~pre4-1 | unstable | alpha, amd64, arm, armel, hppa, hurd-i386, i386, ia64, mips, mipsel, powerpc, s390, sparc The SpanDSP one in Etch is too old and the one in Lenny is too new. That said, I really don't see why people stick with that old spandsp 0.0.4pre16 . There have been bug fixes since. Not to mention some crashes. This is why I bothered backporting spandsp from Asterisk 1.6.0 (app_fax.c is also a better behaving Asterisk module. e.g. does not keep its own personal log files). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - standardization?
On Mon, Dec 22, 2008 at 09:04:13AM -0600, Wesley Haut wrote: Hi all, I know I'm probably stirring up a hornet's nest with this question/comment but I've spent the last few days working on a PHP-based class for the manager interface Isn't there one already? as we're preparing for a pretty big upgrade at our call center and I'm revamping all of the management apps I've written. I can connect to the manager interface and send query strings back and forth all day long, but I'm having issues when it comes to parsing the return data. My issue isn't a PHP one, but rather issues with the consistency of the Manager interface return values. For instance, for the built in actions such as Login, SIPPeers, QueueStatus (the three that I will be calling the most) NONE of those three has a standard return value. Login doesn't return an Event: LoginComplete flag, so the login function can't use the generic Action function I've written. A *Complete even is sent as and end of a series of events. e.g. SIPPeers retuns a separate event for each peer and hence you need an event to tell you that the series is done. For Login, Ping and whatever you know you should not expect further events to follow after the initial reply. SIPPeers returns an additional line after Event: PeerlistComplete, and Peerlist != SIPPeers (meaning the action and eventcomplete flags should match, IMHO). The way QueueStatus returns data is the ideal, the action flag (QueueStatus) has a corresponding complete tag (QueueStatusComplete) and I can (fairly) easily parse the return data by tacking 'Complete' onto the action flag. I just wanted to see what everyone else though of coming up with a standard for ALL manager commands so building applications to hook into Asterisk isn't a crap shoot like it is now. My initial proposal: Any $action has a corresponding $actionComplete event (ie SIPPeers would be SIPPeersComplete instead of PeerlistComplete) The ActionComplete event is the LAST line for any return Like I said, I'm sure this will ruffle some feathers and that is not my intent, maybe I'm missing something here so please (kindlly) inform me if I am. For a larger audince of such proposals, try the -dev list . Be sure to refer to the trunk version. Note, however, that any change to the manager interface may break existing programs. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: You seem to have ommited the relevant links: http://blog.krisk.org/2008/12/introducing-recqual.html http://admin.star2star.com/recqual/ Looks interesting... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?
On Tue, Dec 23, 2008 at 10:35:19AM +0100, Klaus Darilion wrote: Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) users.conf is a hack to generate a typical Asterisk configuration easily. So I figure that patches are welcomed. Assuming that the type won't have any strange ill-effects on any other module that parses that file: A quick grep in trunk gives: apps/app_directory.c apps/app_voicemail.c channels/chan_agent.c channels/chan_dahdi.c channels/chan_h323.c channels/chan_iax2.c channels/chan_sip.c main/manager.c pbx/pbx_config.c res/res_phoneprov.c -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail and Directory not working?
On Mon, Dec 22, 2008 at 03:30:02PM -0500, Noah Miller wrote: Hi All - I'm wondering if anybody has IMAP Voicemail AND the directory working together. I haven't had any success. IMAP voicemail works fine, but when it's active, the Directory does not work. The problem seems to be with libc-client. Specifically, asterisk is not able to access the mm_dlog function. I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu 8.10 and Fedora 9. In each case, I used the native package manager to install libc-client, and in each case, after asterisk is compiled and voicemail users are configured, I get an error in the log that says this: On Ubuntu and Debian (Lenny/Sid) - apt-get source asterisk # as root / using sudo: apt-get build-dep asterisk cd asterisk-1tabtab ASTERISK_NO_DOCS=yes fakeroot debian/rules build Does it build? If so, you have a similar version of Asterisk that builds with IMAP support. (ASTERISK_NO_DOCS=yes is merely intended to save you some unnecessary 5 minutes and ~80MB of generating the API docs) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Thanks all for your replies. I have an aastra 9133i here for testing and am getting a polycom 320 to try out. But today, I got my hands on an older Cisco 7912G with SIP software installed. It connected fine to the Asterisk box, works with the PoE stuff I have, sounds good and doesn't seem to have any problems. Best all, I can buy near new for about $60 each in Australian dollars (thats about 45USD with the Aussie dollar being what it is :) The handsets look OK, they are nice and solid feeling and very easy to use / not complex. Any reason not to use the 7912G ? Seems with the SIP image they work just dandy... Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] regarding query registered or online users fro out of asterisk
Hi all, anyone have any experience regarding query whether our sip accounts registered (online) or not registered (offline) from out of asterisk with mysql or another tool. My goal is taking this information with query and put it to my intranet to check my users. any help would be appreciated Yavuzhan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outging ---asterisk -bug
On Tue, Dec 23, 2008 at 09:06:26AM +0100, Stefan Schmidt wrote: jordan pan schrieb: Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid: 15015895665 [myivr] exten = s,1,Background(test) exten = s,n,WaitExten Where is extension 808 there? What do you see in the CLI trace there? Try: core set verbose 3' beforehand. you should have a look at originate CLI command. you can call using asterisk -rx originate Zap/g0/1501589... Background(test). You meant: asterisk -rx originate Zap/g0/15015895665 application Background test -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
We have one 7912 which we bought for evaluation. The main drawback is that it has hands free speaker but no microphone. __Yehavi: 2008/12/23 Mikel Lindsaar raasd...@gmail.com Thanks all for your replies. I have an aastra 9133i here for testing and am getting a polycom 320 to try out. But today, I got my hands on an older Cisco 7912G with SIP software installed. It connected fine to the Asterisk box, works with the PoE stuff I have, sounds good and doesn't seem to have any problems. Best all, I can buy near new for about $60 each in Australian dollars (thats about 45USD with the Aussie dollar being what it is :) The handsets look OK, they are nice and solid feeling and very easy to use / not complex. Any reason not to use the 7912G ? Seems with the SIP image they work just dandy... Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound fax issues
On Tue, Dec 23, 2008 at 1:28 AM, Danny Nicholas da...@debsinc.com wrote:
What does your extensions.conf look like for this call? If you can
insert a ww into your Dial command (ie, change 18005551212 to ww18005551212)
this may improve your dialing behavior.
In an attempt to isolate the problem, I reduced the extensions.conf to:
; Fax Lines
exten = _.,1,Dial(${AAPT}${EXTEN},,R)
What does putting ww at the front do?
Regards
Mikel
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Re: [asterisk-users] regarding query registered or online users fro out of asterisk
On Tue, Dec 23, 2008 at 5:23 PM, yavuzhan canli yca...@tekfen.com.trwrote: Hi all, anyone have any experience regarding query whether our sip accounts registered (online) or not registered (offline) from out of asterisk with mysql or another tool. My goal is taking this information with query and put it to my intranet to check my users. any help would be appreciated you can use sip show peers console command, or use SIPPees manager Action, in peerentry if you find IPaddress as none that account is offline other wise online. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+SIPpeers -- Godson Gera http://godson.in/voip-asterisk-consultant-hyderabad-india ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound fax issues
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mikel Lindsaar wrote: What does putting ww at the front do? Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial. (It may be a 1/4 second each 'w') Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJUOFkCFu3bIiwtTARAgyWAKCt/LOdJNdL6763OWXP/K3AuHyOJACfZQuO 0kJs1pvYx9rYIoLOys4eQTs= =cTza -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Tue, Dec 23, 2008 at 11:01 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: We have one 7912 which we bought for evaluation. The main drawback is that it has hands free speaker but no microphone. That's true. But we will be getting higher models for the speaker function. Did you find or know of a way to do paging with the Cisco 7912G ? Looking around on Google didn't come up with much. Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound fax issues
On Wed, Dec 24, 2008 at 12:02 AM, Barry L. Kline blkl...@attglobal.netwrote: What does putting ww at the front do? Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial. (It may be a 1/4 second each 'w') I thought so, in that case, it is not the problem here. My problem is that the fax dials out, connects, but can't handle some perceived line noise Need to work out or find out what the best settings are for a fax machine connected to a VOIP device per my diagram previously... Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - standardization?
Isn't there one already? Yeah, but none of them have worked for me...maybe their way of doing things is just different from my approach but I wasn't happy with any of the existing classes. I wasn't planning on releasing my code to the wild (I'm not a programmer by trade I just play one on TV). A *Complete even is sent as and end of a series of events. e.g. SIPPeers retuns a separate event for each peer and hence you need an event to tell you that the series is done. For Login, Ping and whatever you know you should not expect further events to follow after the initial reply. Um, not exactly. I just ran a SIPPeers on the CLI via telnet and each of the PeerEntry events does not have a Complete flag, just the double return. I guess what I was after was a standard way to finalize the larger event as a whole, and keep the existing double line return for separating the individual events. Again, maybe I'm going about this the wrong way as well. For a larger audince of such proposals, try the -dev list . Be sure to refer to the trunk version. Note, however, that any change to the manager interface may break existing programs. Thanks, I'll wait until I've gotten further along on my project before bothering the devs http://iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] second trunk in extensions.conf
I have a TE210P digium card that has 2 E1/T1 ports.
the code in my extensions.conf file for span 1 is :
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1; Trunk interface
TRUNKX=Zap/g2 ; 2nd trunk interface
...
...
; dial a long distance outbound number to SPAIN
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten = _00034X,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _00034X,2,Dial(${TRUNK}/${EXTEN:1},55,To)
exten = _00034X,3,Hangup
...
...
and it works fine, but I need to start working with my second span I don't
know how to add it in extensions.conf file.
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Re: [asterisk-users] Ghost in the Channel-Banks
On Dec 22, 2008, at 10:38 PM, Martin Lima wrote: On Thursday 18 December 2008, Justin Phelps wrote: I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP DiTech Echo Cancel device Voice Channel-Bank (same) T1 from ISP (same) DiTech Echo Cancel device asterisk1 server zap card fax channel bank (same) T1 from ISP (same) DiTech Echo Cancel device asterisk1 server zap card asterisk2 server Now, let me explain the symptoms. d-channel errors on the asterisk1 server on span1 (which is the line coming from the echo cancel from the ISP). asterisk2 server isn't being used as far as I can tell. I've got a red alarm on the port on asterisk1 that asterisk2 is plugged into. I would bet your asterisk2 server was meant for some kind of transition to a different setup. Is there at least some dialplan inside? sip.conf? iax, voicemail etc...? Something does not fit here. If you have the T1 from the ISP going to the echo can, then it cannot go to more than one device. It is not a MUX as far as I remember, have used to great effect in past. Each T1 in matches up to a T1 out. faxes through the fax channel bank are working most of the time. There seems to be problems with multipage faxes. Not isolated to a particular port on the channel bank. As I understand it, echo cancel for faxes is a bad thing, so I don't understand why the previous IT Admin here setup the system as such. Some echo cancellers detect a fax transmission and turn echo cancellation off. In your case if I understand your setup correctly such a device is ambiguous at the best... Possibly it used to be another set of voice lines converted to fax without changes in configuration? I dont really understand the role of echo canceller on E1/T1... Guarantee the Ditech deteects fax and cancels echo can. However any IP in FAX path is problematic for most FAX. If possible turn off ECM. If symptoms are mainly on a single fax machine make sure you have enough memory to buffer entire fax. Some systems let you print as they come in, but I have also seen some which are still trying to buffer while printing and run over. This showed up with a mortgage company doing 50 page legal forms. First 20 or so were fine then started bombing. Best method I have found to troubleshooting FAX is to use a machine which generates a T30 trace output upon completion. Some voice phones on the voice channel banks were not recognizing tones Why the phone should recognize tones? It just generated them while dialing. when dialing. That seems to have been resolved after power cycling the channel bank a few times, and restarting asterisk2 (odd that there doesn't seem to be anything active on asterisk2) Looks like some strange ground loops in your wiring, power issue or something similar. I've been working with the ISP on the d-channel stuff, and things seem to get a little better as they reset equipment, but the d-channel errors have not gone away. Some as above... BTW they will always tell you they reset something no matter what they have really done... :-) Again if this is a PRI, how is signalling being done for 2 T1 connections when only one D Channel. I suspect a partial PRI? I really need advice on these problems. From what I've said, where do you think the problem lies? in the channel banks? in the echo canceller? in the asterisk2 or asterisk1 server? With the ISP? Hard to guess without deeper knowledge of your setup. Intermittent errors and hardware lockups are often caused by power conditions, potential differences and spikes in the powerline. (two pieces of equipment connected together but plugged into two different power outlets coming from two opposite ends of the building can cause real headache!) Check it first. Then you may want to continue what your former IT admin tried to start :-) Are D Channels mostly working just intermittant? I assume you have Ditech set for passthrough on these channels? ie no echo cancel? Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail and Directory not working?
Hi Tzafrir - I'm wondering if anybody has IMAP Voicemail AND the directory working together. I haven't had any success. IMAP voicemail works fine, but when it's active, the Directory does not work. The problem seems to be with libc-client. Specifically, asterisk is not able to access the mm_dlog function. I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu 8.10 and Fedora 9. In each case, I used the native package manager to install libc-client, and in each case, after asterisk is compiled and voicemail users are configured, I get an error in the log that says this: On Ubuntu and Debian (Lenny/Sid) - apt-get source asterisk # as root / using sudo: apt-get build-dep asterisk cd asterisk-1tabtab ASTERISK_NO_DOCS=yes fakeroot debian/rules build Does it build? If so, you have a similar version of Asterisk that builds with IMAP support. I finally got this to work. For some reason, none of the packaged versions of libc-client from any of the distributions I tried support mm_dlog, which is required by the Directory app. I ended up compiling from uw-imap's source on Ubuntu, and that worked right away. On the Red Hat varieties, compiling from source worked, but I had to specify -fPIC and a few other compiler flags when building UW's c-client. For the record, if anybody needs to do this on a redhat platform: 1. Download imap-2007e (or latest version) from ftp://ftp.cac.washington.edu/imap/ 2. Unpack and compile with a make command like: make platform SSLTYPE=none EXTRACFLAGS=-DIGNORE_LOCK_EACCES_ERRORS=1 \ -I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall -Wno-pointer-sign -Wno-parentheses (See the Makefile for a list of platforms - I used 'lr5' for CentOS 5.2) 3. In the asterisk source, run the configure script with the imap flag: ./configure --with-imap=/path/to/imap-source (use the base directory of the imap source - e.g. /usr/src/imap-2007e ) 4. Run make menuselect for asterisk and select IMAP_STORAGE from the Voicemail Build Options. Of course, you'll also need an appropriately configured IMAP server (for CentOS, I recommend their default choice of Dovecot). - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio
Dear Sir, I used several other Softphones like Skype and they are facing the same problem...It seems that the issue is global du to an undersea cable cut Regards On Mon, Dec 22, 2008 at 9:07 PM, michel freiha mich...@gmail.com wrote: Hi all, Sometimes when making a PC to PSTN call through asterisk, I got no audio in both sides...tracing by wireshark, I can find that RTP packets are hitting my PC but no audio...Can someone guess what could be that issue? Maybe it's a latency issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail and Directory not working?
On Tue, Dec 23, 2008 at 10:13:12AM -0500, Noah Miller wrote: Hi Tzafrir - I'm wondering if anybody has IMAP Voicemail AND the directory working together. I haven't had any success. IMAP voicemail works fine, but when it's active, the Directory does not work. The problem seems to be with libc-client. Specifically, asterisk is not able to access the mm_dlog function. I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu 8.10 and Fedora 9. In each case, I used the native package manager to install libc-client, and in each case, after asterisk is compiled and voicemail users are configured, I get an error in the log that says this: On Ubuntu and Debian (Lenny/Sid) - apt-get source asterisk # as root / using sudo: apt-get build-dep asterisk cd asterisk-1tabtab ASTERISK_NO_DOCS=yes fakeroot debian/rules build Does it build? If so, you have a similar version of Asterisk that builds with IMAP support. I finally got this to work. For some reason, none of the packaged versions of libc-client from any of the distributions I tried support mm_dlog, which is required by the Directory app. I ended up compiling from uw-imap's source on Ubuntu, and that worked right away. http://packages.ubuntu.com/intrepid/asterisk http://packages.ubuntu.com/intrepid/amd64/asterisk/filelist app_voicemail_imap.so is included . I built several newer versions on my Lenny system and never encountered such an issue. Hmm but in my 1.6.0 copy app_voicemail_imap.so doesn't have mm_dlog as well. It came from http://svn.digium.com/svn/asterisk/branches/1...@134223 (Merging the imap_consistency branch.) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?
On Tue, Dec 23, 2008 at 4:40 AM, Steve Totaro stot...@first-notification.com wrote: It's all ball bearings these days What is the deal with Fletch quotes these days? Don't get me wrong, I appreciate them but I'm starting to wonder where this is all coming from. I *think* it's because Fletch has been on HBO lately. Am I correct? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 6:31 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: You seem to have ommited the relevant links: http://blog.krisk.org/2008/12/introducing-recqual.html http://admin.star2star.com/recqual/ Looks interesting... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir Tzafrir, I included a link to my blog in the original post but it probably wasn't as clear as it could have been. Looks interesting? Thanks! I'd love to see it deployed more, I've only used it in a few specific scenarios. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pattern Matching
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is present, replace it with Unknown. I'm using the
ael format for my dialplan and have been looking for a way to do this,
but haven't found anything yet. Is there a way to do this inside the
dialplan or do I have to pass it out to an AGI script?
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[asterisk-users] Dailplan code for holiday detection?
This has been on my ToDo list far too long. I have a small call-center setup, with basic time of day/day of week validation before putting callers in the queues. With the holidays upon us, I need to add check to see if 'today' is a holiday so I do not put callers in unmanned queues. Due to how the agents work, I have to allow joinwhenempty. Does anyone have a snippet of dialplan code, perhaps using Astdb, to check it 'today' is a listed holiday? Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL Variable Warning Messages
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
defined 2 channel groups in my zapata.conf. The second office needs all
of their outgoing calls to go out over their lines so the people they
call will have the correct callerID. I created an asterisk database and
with entries in the database for all extensions in the second office and
defined the following macro:
globals {
CONSOLE=Console/dsp;
TRUNK=Zap/r1;
TCTC_Operator=15;
Law_Operator=12;
};
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
It's working and correctly routing outside calls, but I get the
following messages when I reload the extensions.ael file:
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
Any idea what is causing the warnings?
Thanks,
Brent
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Re: [asterisk-users] Pattern Matching
On Tuesday 23 December 2008 11:47:08 Brent Davidson wrote:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is present, replace it with Unknown. I'm using the
ael format for my dialplan and have been looking for a way to do this,
but haven't found anything yet. Is there a way to do this inside the
dialplan or do I have to pass it out to an AGI script?
Set(CALLERID(name)=${IF($[${CUT(CALLERID(name),/,1)}=Zap]?Unknown:
${CALLERID(name)})})
--
Tilghman
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Re: [asterisk-users] Pattern Matching
Brent Davidson schrieb:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is present, replace it with Unknown. I'm using the
ael format for my dialplan and have been looking for a way to do this,
but haven't found anything yet. Is there a way to do this inside the
dialplan
if (${CALLERID(name):0:4} = Zap/) {
Set(CALLERID(name)=Unknown);
}
Not sure why you would want to put the channel name into the
caller ID name in the first place.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Dailplan code for holiday detection?
On Tuesday 23 December 2008 12:11:41 Dan Austin wrote:
This has been on my ToDo list far too long.
I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.
With the holidays upon us, I need to add check to
see if 'today' is a holiday so I do not put callers
in unmanned queues. Due to how the agents work, I have
to allow joinwhenempty.
Does anyone have a snippet of dialplan code, perhaps using
Astdb, to check it 'today' is a listed holiday?
Astdb is a nice idea. Something along the lines of:
GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)
would work. Holidays are evaluated as 01, which is true. Anything not
in the database would be evaluated as 0, which is false. This will work
both for holidays where the date changes every year (e.g. Thanksgiving,
Labor Day), as well as holidays where it doesn't (e.g. Christmas, Independence
Day).
--
Tilghman
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Re: [asterisk-users] AEL Variable Warning Messages
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Pattern Matching
Philipp Kempgen wrote:
Brent Davidson schrieb:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is present, replace it with Unknown. I'm using the
ael format for my dialplan and have been looking for a way to do this,
but haven't found anything yet. Is there a way to do this inside the
dialplan
if (${CALLERID(name):0:4} = Zap/) {
Set(CALLERID(name)=Unknown);
}
Not sure why you would want to put the channel name into the
caller ID name in the first place.
Philipp Kempgen
Thanks all. As far as why the channel name is in the caller ID, I don't
know. I'm certainly not doing it intentionally. I don't have any code
in the dialplan that even touches the CallerID, so I guess Asterisk is
doing somehow when the Name part of the CallerID is unknown... Either
that or my Snom 300 phones are picking the wrong info to use for CallerID.
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Re: [asterisk-users] AEL Variable Warning Messages
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);
Philipp Kempgen
According to the AEL Documentation I should be able to set variables
without using the Set command. They even give the following example:
context foo {
555 = {
x=5;
y=blah;
divexample=10/2
NoOp(x is ${x} and y is ${y} !);
};
};
I wonder if maybe AEL is ignoring the double quotes and treating the
Zap/r2 as if it were division??? Should I file a bug report on this?
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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: Looks very interesting. After reading all available info I have two questions before testing: 1. Who/what answers the calls at the other end? I guess real live traffic should be sent through this Asterisk server? 2. How many calls you had made to to diagnose your problems? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Brent Davidson wrote:
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);
Philipp Kempgen
According to the AEL Documentation I should be able to set variables
without using the Set command. They even give the following example:
context foo {
555 = {
x=5;
y=blah;
divexample=10/2
NoOp(x is ${x} and y is ${y} !);
};
};
I wonder if maybe AEL is ignoring the double quotes and treating the
Zap/r2 as if it were division??? Should I file a bug report on this?
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 2:59 PM, Mindaugas Kezys mke...@gmail.com wrote: Looks very interesting. After reading all available info I have two questions before testing: 1. Who/what answers the calls at the other end? I guess real live traffic should be sent through this Asterisk server? 2. How many calls you had made to to diagnose your problems? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions Mindaugas, Thanks. 1) The same Asterisk system places the call and answers the call. You can use one technology to place the call and another to answer (but you don't have to). One connection is the control, the other is the variable. For instance, in my experiments I was testing SIP providers against one another and each other. I placed outbound calls to the SIP provider to a DID I had on a local PRI that came back into the same system running recqual. Recqual then (as designed) played audio in both directions while recording, like this: outbound call (to SIP provider, dialing DID on PRI below): - (audio in from inbound leg) -- (audio out to outbound leg - not recorded, should always be perfect) Inbound call (from PRI): - (audio in from outbound leg) -- (audio out to inbound leg - not recorded, should always be perfect) I know what the audio on both legs should look like so all I need to do is analyze it once it is returned to me. That way I can see how it changed from what Asterisk was supposed to be generating in the first place. This is how recqual is able to detect such a wide range of quality problems. 2) Depends on what you are testing for. For example, if you know you have a bad pair in a bundle of analog lines, you could just call all of the numbers a couple of times. Whichever number/pair has the worst audio (fails the test or looks the worst) is the bad pair. In the case of my SIP provider, they used outbound load balancing to send traffic to multiple media gateways and multiple trunk groups all over the country. Because we had no control over these routing decisions we had to make periodic calls for at least a day to detect the majority of the bad hosts/trunks/etc. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Directory exists when * is pressed....but where?
I have been trying to figure out how the * works when in the Directory (dial-by-name). When I press * (which is supposed to exit the directory) I end up somewhere which I never specified. It seems like Asterisk just picked that place to go, because I never specified it. The wiki is no help on that one Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directory exists when * is pressed....but where?
On Tuesday 23 December 2008 14:49:52 Mike wrote: I have been trying to figure out how the * works when in the Directory (dial-by-name). When I press * (which is supposed to exit the directory) I end up somewhere which I never specified. It seems like Asterisk just picked that place to go, because I never specified it. When you press '*', it enters the 'a' extension in the dial context (second argument to Directory, or first, if the second is not specified). If the 'a' extension does not exist, then Directory exits normally and the next priority in the current extension is executed. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Mon, Dec 22, 2008 at 5:37 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: Recqual is collection of scripts using Asterisk and other Linux utilities to measure call quality on an automated basis. -How it Works- Recqual was designed to detect audio quality problems in a call path that may not be visible from the technology being used locally. Whether it's SIP, ZAP, or IAX the fact is there are many potential sources of call quality problems in just about any call being made. Often times a SIP provider may resell services, for example. While the delivery of IP packets to/from this provider may look excellent there may be other problems upstream that an analysis of the IP packets, path, etc may not be able to detect. In scenarios such as this the only way to identify call quality problems is to analyze the audio itself. Regardless of method or transport being used, the goal of any telephony system is to deliver reliable, consistent call quality. Recqual is designed to allow you to place a large number of automated calls (using Asterisk) using different call scenario files. The key here is consistency. When Asterisk places the outbound call (and answers the inbound call) it will generate a set of tones while recording the return audio path. Once the run has finished Ecasound will run with various filters and noise gates to detect certain amounts of distortion, signal loss, etc. Calls either pass or fail based on how much variation there is in the audio once it has been returned. Of course you can pass audio through any combination of networks - including the PSTN. Almost any call quality problem(s) can be detected with this method. Whether it's one way calls, echo, dropped packets, distortion, etc ecasound should be able to isolate the problem calls. If not you can just tweak the script ;). Only calls that fail are saved. These files can be imported into your favorite audio processing utility and/or run through Ecasound again if you'd like to tweak the process script to detect them automatically. Recqual has been designed (and optimized) to work with SIP channels. For example, it has the ability to correlate problem calls with specific RTP endpoint IP addresses. However, due to the protocol independent nature of Asterisk you can use just about any channel type with a few simple changes. --- So there you have it. I've used this with a great deal of success but I think there is still a lot to be done. More on my blog here: http://blog.krisk.org Thoughts? Hi, This is good idea, and i will probably try it out someday next year (too busy completing my business requirements :) I took a look at asterisk patch, and it seems quite simple. I just don't see the point of removing if(debug). You could easily get this additional logging into Asterisk trunk (if preserving RTP info in debug level), and starting asterisk with debug 1. So, then it would be easier to install recqual. Also, being able to run on unmodified version of Asterisk, it would be good to allow keeping current dialplan and just route test calls trough it. So, people would be able to keep track of their billing, etc for those test calls. Also, thanks for showing us magics of ecasound. I have similar project (pbx-test-framework) that allows IVR/Queue/etc testing in automated mode. Recording everything and checking voice quuailty would be great addition :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directory exists when * is pressed....but where?
Mike wrote: I have been trying to figure out how the * works when in the Directory (dial-by-name). When I press * (which is supposed to exit the directory) I end up somewhere which I never specified. It seems like Asterisk just picked that place to go, because I never specified it. The wiki is no help on that one… Mike If you look at the help text for the Directory application using the Asterisk CLI (core show application directory), it specifies that pressing the '*' key will send you to the 'a' extension if it exists. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Dave Fullerton wrote: I had gotten similar messages when I forgot to put quotes around channels like that (took me forever to realize that one). Since you have them I would say this is a bug. What version of asterisk are you running? -Dave I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with DAHDI and Rhino RCBFX card. I tried doing a new install with 1.4.22 yesterday and couldn't get Oslec to work correctly with the Rhino card when running with DAHDI instead of zaptel. Unfortunately 1.4.22 no longer has Zaptel. :( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
On Tue, 23 Dec 2008, Brent Davidson wrote: Dave Fullerton wrote: I had gotten similar messages when I forgot to put quotes around channels like that (took me forever to realize that one). Since you have them I would say this is a bug. What version of asterisk are you running? -Dave I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with DAHDI and Rhino RCBFX card. I tried doing a new install with 1.4.22 yesterday and couldn't get Oslec to work correctly with the Rhino card when running with DAHDI instead of zaptel. Unfortunately 1.4.22 no longer has Zaptel. :( Why do you need oslec to work with the rhino card - it has hardware echo cancellation built in doesn't it? j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dailplan code for holiday detection?
Not the most elegant but since I have a generic context for my IVRs I
simple check the date there.
exten = s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|3|jul?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|7|sep?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|26|nov?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|27|nov?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|dec?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|26|dec?closed-holiday|1)
exten =
closed-holiday,1,Background(ivr-closed-holiday-${AUTOATTENDANT}||)
exten = closed-holiday,n,Hangup
This is next year's holidays for us but with this year's Christmas days
in it.
sl
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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 4:02 PM, Atis Lezdins a...@iq-labs.net wrote: Hi, This is good idea, and i will probably try it out someday next year (too busy completing my business requirements :) Luckily next year is just over a week away. We won't have to wait that long ;). I took a look at asterisk patch, and it seems quite simple. I just don't see the point of removing if(debug). You could easily get this additional logging into Asterisk trunk (if preserving RTP info in debug level), and starting asterisk with debug 1. So, then it would be easier to install recqual. Also, being able to run on unmodified version of Asterisk, it would be good to allow keeping current dialplan and just route test calls trough it. So, people would be able to keep track of their billing, etc for those test calls. This is true, however, I wasn't very excited about any other debug messages that might get printed with debug 1. I knew I only needed the endpoint RTP address, so I just removed the if. Of course you could always just run with debug 1 instead of the patch too. Again, this modification isn't strictly required. I just did if for SIP providers that give unpredictable media endpoint IP addresses... :) Also, thanks for showing us magics of ecasound. I have similar project (pbx-test-framework) that allows IVR/Queue/etc testing in automated mode. Recording everything and checking voice quuailty would be great addition :) Ecasound is very, very cool. Recqual is only scratching the surface of what it can do! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote: Unfortunately 1.4.22 no longer has Zaptel. :( Asterisk 1.4.22 builds with both Zaptel and DAHDI. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 11:17 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: This is true, however, I wasn't very excited about any other debug messages that might get printed with debug 1. I knew I only needed the endpoint RTP address, so I just removed the if. Of course you could always just run with debug 1 instead of the patch too. Again, this modification isn't strictly required. I just did if for SIP providers that give unpredictable media endpoint IP addresses... :) Debug 1 isn't that much. Just grep for line you're using and everything should work fast and fine. Sometimes i even log our production servers for weeks with debug 1. So i would suggest submiting this modification to digium bugtracker, if it really helps tracking ip's. Thanks again, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Tzafrir Cohen wrote: On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote: Unfortunately 1.4.22 no longer has Zaptel. :( Asterisk 1.4.22 builds with both Zaptel and DAHDI. I spent several hours trying to make it work yesterday and it just wouldn't. I kept getting an error message that it was unable to bind the echo canceler to channel 1. It might have something to do with the RCBFX drivers, I'm not sure. I found your page and followed your instructions. Everything appeared to work until I checked with dahdi_cfg -vv. That's where I got the message. Don't have my notes here so I don't have the actual error message right now. -- Brent Davidson I.T. Manager Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directory exists when * is pressed....but where?
Thanks, to you and Mark, for the quick reply. I used to rely on the Wiki but it seems I shouldn't Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, December 23, 2008 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory exists when * is pressedbut where? On Tuesday 23 December 2008 14:49:52 Mike wrote: I have been trying to figure out how the * works when in the Directory (dial-by-name). When I press * (which is supposed to exit the directory) I end up somewhere which I never specified. It seems like Asterisk just picked that place to go, because I never specified it. When you press '*', it enters the 'a' extension in the dial context (second argument to Directory, or first, if the second is not specified). If the 'a' extension does not exist, then Directory exits normally and the next priority in the current extension is executed. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Jeff LaCoursiere wrote: On Tue, 23 Dec 2008, Brent Davidson wrote: Dave Fullerton wrote: I had gotten similar messages when I forgot to put quotes around channels like that (took me forever to realize that one). Since you have them I would say this is a bug. What version of asterisk are you running? -Dave I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with DAHDI and Rhino RCBFX card. I tried doing a new install with 1.4.22 yesterday and couldn't get Oslec to work correctly with the Rhino card when running with DAHDI instead of zaptel. Unfortunately 1.4.22 no longer has Zaptel. :( Why do you need oslec to work with the rhino card - it has hardware echo cancellation built in doesn't it? j The Rhino card is supposed to have hardware echo cancellation. That's one of the main reasons I switched to that card from the X-100p's I was using. Unfortunately, either I don't know how to turn on the hardware echo cancellation or it just doesn't work. I have 5 separate location where I'm using that card and if I turn off Oslec at any of them the echo is so bad that the systems is virtually unusable. With Oslec enabled, however, there is no echo at all. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dailplan code for holiday detection?
We chose to use a mySQL database to store the holiday information.
When a call is answered we query the database to see if there is a
holiday greeting recorded, if so we play the indicated greeting,
otherwise play the default menu greeting. (We do our dialplans in AEL)
context checkHoliday {
s =
{
begin:
MYSQL(Connect temp communicator username password
asterisk);
MYSQL(Query resultid ${temp} SELECT greeting FROM
menuGreetings WHERE startTime=FROM_UNIXTIME(${EPOCH}) AND
endTime=FROM_UNIXTIME(${EPOCH}) LIMIT 1);
MYSQL(Fetch foundRow ${resultid} sqlGreeting);
MYSQL(Clear ${resultid});
MYSQL(Disconnect ${temp});
if (${foundRow}==1)
{
Background(custom/mainMenu/${sqlGreeting});
goto mainMenu,s,begin;
}
else
{
goto checkTime,s,begin;
}
}
includes
{
mainMenu;
tempGreeting;
voicemail;
publicExt;
}
};
The 'checkTime' context simply checks if we are open or closed and
plays the appropriate greeting (if no holiday greeting is found).
Daniel
On Dec 23, 2008, at 1:14 PM, Scott L. Lykens wrote:
Not the most elegant but since I have a generic context for my IVRs I
simple check the date there.
exten = s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|3|jul?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|7|sep?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|26|nov?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|27|nov?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|dec?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|26|dec?closed-holiday|1)
exten =
closed-holiday,1,Background(ivr-closed-holiday-${AUTOATTENDANT}||)
exten = closed-holiday,n,Hangup
This is next year's holidays for us but with this year's Christmas
days
in it.
sl
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Re: [asterisk-users] AEL Variable Warning Messages
On Tue, Dec 23, 2008 at 04:01:24PM -0600, Brent Davidson wrote: Tzafrir Cohen wrote: On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote: Unfortunately 1.4.22 no longer has Zaptel. :( Asterisk 1.4.22 builds with both Zaptel and DAHDI. I spent several hours trying to make it work yesterday and it just wouldn't. I kept getting an error message that it was unable to bind the echo canceler to channel 1. What error message from where? With Zaptel the echo canceller settings are global (that is: one hard-coded echo canceller). With DAHDI there are echo canceller modules and you can (and actually need to) set them per-channel. It might have something to do with the RCBFX drivers, I'm not sure. I found your page and followed your instructions. Everything appeared to work until I checked with dahdi_cfg -vv. That's where I got the message. Don't have my notes here so I don't have the actual error message right now. -- Brent Davidson I.T. Manager Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dailplan code for holiday detection?
Tilghman wrote:
Astdb is a nice idea. Something along the lines of:
GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)
would work. Holidays are evaluated as 01, which is true.
Anything not in the database would be evaluated as 0, which
is false. This will work both for holidays where the date
changes every year (e.g. Thanksgiving, Labor Day), as well as
holidays where it doesn't (e.g. Christmas, Independence Day).
On one hand I am embarrassed that it is that simple, on the other
I am thrilled that it is that simple.
After the Holidays I guess I need to put together a cheesy
web page to allow for adding the dates to Astdb, but for now
this is awesome and much appreciated.
Thanks,
Dan
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Re: [asterisk-users] AEL Variable Warning Messages
Tzafrir Cohen wrote: What error message from where? With Zaptel the echo canceller settings are global (that is: one hard-coded echo canceller). With DAHDI there are echo canceller modules and you can (and actually need to) set them per-channel. It might have something to do with the RCBFX drivers, I'm not sure. I found your page and followed your instructions. Everything appeared to work until I checked with dahdi_cfg -vv. That's where I got the message. Don't have my notes here so I don't have the actual error message right now. I don't remember the actual error name, but it showed up when I did dahdi_cfg -vv. It was something like DAHDI_ATTACH_ECHO_CANCELLER Failed for Channel 1. Unsupported command (22). I was trying to see if maybe it was logged to my syslog but this is all I find in my /var/log/messages: Dec 22 17:01:43 localhost modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/2.6.23.8x86_64/dahdi/dahdi_echocan_osle c.ko): Unknown symbol in module, or unknown parameter (see dmesg) Dec 22 17:01:43 localhost kernel: rcbfx 1: Spotted a Rhino: Rhino RCB4FXO (4 channels) Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol oslec_create Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol oslec_update Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol oslec_free Also, when using the Dahdi/Oslec/RCBFX combination I was getting tons of blocks like this in my syslog: Dec 22 16:54:58 localhost kernel: 80c = 2c7e5000 Dec 22 16:54:58 localhost kernel: 810 = 240 Dec 22 16:54:59 localhost kernel: 814 = 0 Dec 22 16:55:00 localhost kernel: 818 = 0 Dec 22 16:55:00 localhost kernel: 81c = 0 Dec 22 16:55:00 localhost kernel: 820 = Dec 22 16:55:01 localhost kernel: 824 = Dec 22 16:55:01 localhost kernel: 828 = Dec 22 16:55:02 localhost kernel: 82c = 0 Dec 22 16:55:02 localhost kernel: 830 = Dec 22 16:55:02 localhost kernel: 834 = Dec 22 16:55:02 localhost kernel: 838 = Dec 22 16:55:03 localhost kernel: 83c = 0 Dec 22 16:55:03 localhost kernel: 840 = 3 Dec 22 16:55:04 localhost kernel: 844 = f Dec 22 16:55:04 localhost kernel: 848 = Dec 22 16:55:04 localhost kernel: 84c = 0 Dec 22 16:55:04 localhost kernel: 850 = 0 Dec 22 16:55:05 localhost kernel: 854 = 10f Dec 22 16:55:05 localhost kernel: 858 = 14e00ff Dec 22 16:55:12 localhost kernel: 85c = 3d434310 Dec 22 16:55:13 localhost kernel: 860 = 0 Dec 22 16:55:13 localhost kernel: 864 = 0 Dec 22 16:55:18 localhost kernel: 868 = 229e229e Dec 22 16:55:18 localhost kernel: 86c = 0 Dec 22 16:55:19 localhost kernel: 870 = 5 Dec 22 16:55:19 localhost kernel: 874 = 5 Dec 22 16:55:20 localhost kernel: 878 = Dec 22 16:55:20 localhost kernel: 87c = 0 Dec 22 16:55:21 localhost kernel: 880 = 0 Dec 22 16:55:21 localhost kernel: 884 = 0 Dec 22 16:55:21 localhost kernel: 888 = 0 Dec 22 16:55:22 localhost kernel: 88c = 0 Dec 22 16:55:22 localhost kernel: 890 = 0 Dec 22 16:55:22 localhost kernel: 894 = 0 Dec 22 16:55:24 localhost kernel: 898 = 0 Dec 22 16:55:26 localhost kernel: 89c = 0 Dec 22 16:55:28 localhost kernel: 8a0 = 0 Dec 22 16:55:28 localhost kernel: 8a4 = 0 Dec 22 16:55:28 localhost kernel: 8a8 = 0 Dec 22 16:55:28 localhost kernel: 8ac = 0 Dec 22 16:55:28 localhost kernel: 8b0 = 0 Dec 22 16:55:28 localhost kernel: 8b4 = 0 Dec 22 16:55:28 localhost kernel: 8b8 = 0 Dec 22 16:55:28 localhost kernel: 8bc = 0 Dec 22 16:55:28 localhost kernel: 8c0 = 0 Dec 22 16:55:28 localhost kernel: 8c4 = 0 Dec 22 16:55:28 localhost kernel: 8c8 = 0 Dec 22 16:55:28 localhost kernel: 8cc = 0 Dec 22 16:55:28 localhost kernel: 8d0 = 0 Dec 22 16:55:28 localhost kernel: 8d4 = 0 Dec 22 16:55:28 localhost kernel: 8d8 = 0 Dec 22 16:55:28 localhost kernel: 8dc = 0 Dec 22 16:55:28 localhost kernel: 8e0 = 0 Dec 22 16:55:28 localhost kernel: 8e4 = 0 Dec 22 16:55:28 localhost kernel: 8e8 = 0 Dec 22 16:55:28 localhost kernel: 8ec = 0 Dec 22 16:55:29 localhost kernel: 8f0 = 0 Dec 22 16:55:29 localhost kernel: 8f4 = 0 Dec 22 16:55:29 localhost kernel: 8f8 = 0 Dec 22 16:55:29 localhost kernel: 8fc = 0 Dec 22 16:55:29 localhost kernel: 900 = 0 Dec 22 16:55:29 localhost kernel: 904 = 0 Dec 22 16:55:29 localhost kernel: 908 = 0 Dec 22 16:55:29 localhost kernel: 90c = f0f0f0f Dec 22 16:55:29 localhost kernel: 910 = f0f0f0f Switching back to Zaptel solved all of the problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI error
[Dec 23 17:58:49] ERROR[3091]: chan_dahdi.c:8413 dahdi_pri_error: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX I am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7 I am using a TE120P card. I am also getting this VERY frequently: -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'DAHDI/0-1' Versus a normal hangup: == Spawn extension (smvoice-dialout, smvoice_callprogress, 4) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' system.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-3 dchan=24 chan_dahdi.conf: [channels] switchtype=national signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-3 I was using zaptel 1.4.12.1 before using DAHDI and was getting the same thing. Sometimes is appears to work fine - other times I get these SPURIUOS hangups. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directory exists when * is pressed....but where?
On Dec 23, 2008, at 4:49 PM, Mike wrote: Thanks, to you and Mark, for the quick reply. I used to rely on the Wiki but it seems I shouldn't Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, December 23, 2008 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory exists when * is pressedbut where? On Tuesday 23 December 2008 14:49:52 Mike wrote: I have been trying to figure out how the * works when in the Directory (dial-by-name). When I press * (which is supposed to exit the directory) I end up somewhere which I never specified. It seems like Asterisk just picked that place to go, because I never specified it. When you press '*', it enters the 'a' extension in the dial context (second argument to Directory, or first, if the second is not specified). If the 'a' extension does not exist, then Directory exits normally and the next priority in the current extension is executed. -- Tilghman Anyone have issues with this and 1.6.0.rc3? Behavior is repetition of menu on * or 0, not using the o or a variables. Fred Posner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
2008/12/3 Tilghman Lesher tilgh...@mail.jeffandtilghman.com On Tuesday 02 December 2008 12:22:16 Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? I'll get a new release candidate out either this afternoon or tomorrow; I'm currently working on ensuring that 1.6.0.3 will not be a regression from 1.4.23. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I'm using centos 5.2 and I've successfully compiled asterisk 1.6.0.1 , but when I want to compile 1.6.0.2. I get : [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Have you solved? -- Pagarbiai / Best Regards, Giedrius Augys ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
