Re: [asterisk-users] zlib1g
Tzafrir Cohen My advice: If the information is outdated Submit updated information Best regards, Al Bochter Bochter Services -- Need to call use our web phone at the link below http://www.bochterservices.com/voip/iaxphone.php?cn=250 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- Tzafrir Cohen wrote: On Wed, Jun 20, 2007 at 03:32:19PM -0700, bilal ghayyad wrote: Dear Cohen; In this link: http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html In the subject: 2.Installation, then in the sub title: Zaptel Installation Please advise. My advice: don't use obsolete doucmentation. That incorrect recommenndation is not the only mistake in that page. ___ --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] mISDN problems
Hi all, we're buildin an Asterisk box based on an Intel IXP425 board. The board uses a Beronet BN2S0 ISDN card, mISDN 1.1.4 and asterisk 1.4.2. hfc_multi has been patched to compile under big endian cpu, and so also capi kernel files. All the modules seem to load correctly (configuration was made with misdn-init config), but when starting cha_misdn, asterisk outputs the following lines: P[ 1] Restarting this port. P[ 1] Stack:0x174f10 P[ 1] empty_chan_in_stack: 1 P[ 1] $$$ CLEANUP CALLED pid:0 P[ 1] empty_chan_in_stack: 2 P[ 1] $$$ CLEANUP CALLED pid:0 P[ 1] empty_chan_in_stack: 3 P[ 1] $$$ CLEANUP CALLED pid:0 P[ 1] L1: PH L1Link Up! P[ 0] MGMT: SSTATUS: L1_ACTIVATED P[ 1] % GOT L2 DeActivate Info. P[ 1] !!! Could not Get the L2 up after 3 Attemps!!! P[ 1] % GOT L2 Activate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] % GOT L2 DeActivate Info. P[ 1] !!! Could not Get the L2 up after 3 Attemps!!! This error is generated by a function into asterisk package in file channels/misdn/isdn_lib.c misdn-init start output: dip01:/mnt/externfs/beronet/install-misdn-mqueue/mISDN-1_1_4# misdn-init start - Loading module(s) for your misdn-cards: - /sbin/modprobe --ignore-install hfcmulti type=0x1 protocol=0x12,0x22 layermask=0x3,0xf poll=128 debug=0x88 /sbin/modprobe mISDN_dsp debug=0x0 options=0 poll=160 dtmfthreshold=100 dmesg related output: Modular ISDN Stack core version (1_1_4) revision ($Revision: 1.40 $) mISDNd: kernel daemon started (current:c2c2bac0) ISDN L1 driver version 1.20 mISDNd: test event done ISDN L2 driver version 1.32 mISDN: DSS1 Rev. 1.47 mISDN Capi 2.0 driver file version 1.21 mISDN: HFC-multi driver Rev. 1.68 HFC-multi: card manufacturer: 'Cologne Chip AG' card name: 'HFC-2S Beronet Card' clock: double PCI: enabling device :00:05.0 ( - 0003) HFC-2S#1: defined at IOBASE 0x1000 IRQ 28 HZ 100 leds-type 3 HFC_multi: resetting HFC with chip ID=0xc revision=1 hfcpci_probe: DIPs(0x9f) jumpers(0x1) HFC_manager: channel 2 (0..31) data c30d prim f1681 arg HFC_manager: MGR_REGLAYER HFC_manager: channel 2 (0..31) data c30d prim f1482 arg HFC_manager: MGR_SETSTACK HFC_manager: channel 2 (0..31) data c30d prim f4182 arg HFC_manager: channel 6 (0..31) data c63b2800 prim f1681 arg HFC_manager: MGR_REGLAYER HFC_manager: channel 6 (0..31) data c63b2800 prim f1a82 arg cb150e50 HFC_manager: MGR_***STPARA HFC_manager: channel 6 (0..31) data c63b2800 prim f1a82 arg cb150e50 HFC_manager: MGR_***STPARA HFC_manager: channel 6 (0..31) data c63b2800 prim f1482 arg HFC_manager: MGR_SETSTACK 1 devices registered HFC_manager: channel 6 (0..31) data c63b2800 prim f4182 arg mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) mISDN_dsp: DSP clocks every 160 samples. This equals 2 jiffies. The only output that we see when interacting with a phone connected to a PBX is a string like 0x64 0x7f 0x01 but it seems more related to layer1. Once we have seen an error from mISDN_read, coded 22, but never have been able to reproduce it. The same configuration on x86 work perfectly. Any idea? Regards 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
[asterisk-users] SIP/IAX2 Phones behind USR 9108 Router
Hello all, Im having an asterisk server hosted in US. I have extension to the PBX behind a U.S.Robotics Maxg (9108) router. I do get registered on SIP and IAX which i connect the phones but after around 24 hours the registration fails and cant receive calls. Once i restart the router it works again. I tried various methods to make it ok but failed, Some of them were Registration time out to 180s etc. Does anyone has the same problem or am i doing some mistakes.. Pls guide. Thanks Danny ___ --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 config files and #include
Hi all, I am using asterisk version 1.2.18. I recently tried to change my asterisk configuration by using #include statements to include other config files in my extensions.conf and queues.conf files. My queues.conf is in /etc/asterisk. It includes several files which are in /etc/asterisk/queues. Each of these files contains the config of individual queues. Again each of the individual queue config files in /etc/asterisk/queues includes files which are in /etc/asterisk/queues/queue_members. The problem is that when I reload this config I get the following error: - *WARNING: Maaximum include level exceeded : 10* Has anyone encoutered this before and does anyone know what it means ?? Any help will be deeply appreciated as I have been unable to find any documentation on this. Thanks -Deepak ___ --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] Problem with Remote-Hold/MusicOnHold
Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends Remote hold message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to extension 20. But Asterisk doesn't send Remote retrieve, so the mosic-on-hold of the carrier doesn't stop. You can hear the person on Mobile A, but the person on Mobile A only gets music. Anybody with the same issue? It happens on a bristuffed Asterisk 1.2.19. Is there a way to stop sending Remote hold or a way to send Remote retrieve? See here the the cli output with pri debug span 4 at starting attended transfer: 4 Protocol Discriminator: Q.931 (8) len=7 4 Call Ref: len= 1 (reference 214/0xD6) (Terminator) 4 Message type: NOTIFY (110) 4 [27 01 f9] 4 Notification indicator (len= 3): Ext: 1 Remote hold (121) -- Started music on hold, class 'default', on channel 'Zap/10-1' -- Playing 'pbx-transfer' (language 'de') Best regards, Gunnar Schaller ___ --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] gtalk - no audio
Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage=Talk to me timeout=100 *gtalk.conf:* [general] context=default allowguest=yes bindaddr=172.25.123.18 [guest] disallow=all allow=ulaw context=gtalk This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. I first had similar problems, because I didn't use bindaddr in gtalk.conf. But that fixed it for me, but not for most other cases. Also, we all use the same network (same routing and NAT) and Gtalk version. Audio calls between regular Gtalk users is not a problem. This problem really puzzles me. Is it a channel gtalk problem, or do we need to look at other settings (network, client settings...)? I personnaly think we can rule out network config, since both successful and unsuccessful users work in the same lan. Is there anybody with experience in using channel gtalk? Should we start debugging? What can we learn from jabber debug logs? Any help is very much appreciated! koenvi ___ --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] chan problem
Thx, However it appears to be something else. Still need to find out what it is. Loading during boot does not work. After unloading (rmmod) modules mISDN, zaptel, wctdm etc, then reloading them manually in any particular order it works. On Mon, Jun 18, 2007 at 12:36:10PM -0400, Bob Chiodini wrote: [EMAIL PROTECTED] wrote: I experienced the same problem. The only way I could get both ISDN and analog working was unloading kernel modules for zaptel and mISDN after boot and then load them in the order: zaptel first and then mISDN. Still need to find out how to configure load order in linux. Hans, Have a look at the man page for modprobe.conf, specifically the install directive. There is an example of how to force the order. But it is already heavily abused. You may actually want to load one and not the other, and with that directive you can't . One alternative guess is the need to blacklist a third module. -- 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 ___ --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] ipv6 on Asterisk
On 20 Jun 2007, at 10:06, Chris Hills wrote: Jason Ma wrote: Hi guys, Does anybody try to install IPV6 support on asterisk?I just found a patch for that but it is released on 2005,I have no idea if there is new version to support ipv6 or new patches,please advise.Thanks a lot. It is a very desirable feature that will solve a lot of problems, but for one reason or another it has largely been ignored. Actually there has been a lot of work done on it (not by me) There was a session on ipV6 at the recent developer's meeting. However, I'm curious to know which problems you have that will be solved by V6. Tim Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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 GUI
For example, a user could post a message to the list asking I'm new to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX, or buy a commercial solution? Imagine the response as you tried to convince them to buy PBXWare, FreePBX users try to convince them that they should start out using FreePBX, and others go on about how hand coding a dialplan is the one-true-wayR to learn Asterisk. Generally, the original poster is just looking to get everyone stirred up over nothing. In other words, Paul's original post of GUI bad! CLI good! was just the sort of post that is going to get folks fired up re-re-restarting the age-old discussion of which is better: CLI or GUI. Basically, it could be like posting any of the following: - Which is better: emacs or vi? - Which linux distribution is the best? - Which is better: Macs or Windows? All of these questions share the following: 1.) They have no right answer (macs are better for some, Windows for others, and linux for others still, not to mention OS/2, BSD, etc) 2.) People on the various sides of the debate have extremely strong feelings on the matter 3.) Nobody is likely to be convinced that the other side is right and that they are wrong. 4.) They have all been discussed thousands of times before, and nothing new is likely to be said on the matter. 5.) The only purpose served by the discussion, due to the reasons above, is to clutter up the mailing list. 6.) Any discussion thread regarding these sorts of topics is best avoided. For a more thorough description of an internet troll, see the following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 28internet%29 In other words, if you see a post that is just going to result in a re-rehashing of the last rehash of a specific subject, just hit the delete key instead of clogging up the mailing list with yet another thread on whether a GUI or a CLI is better. (for example). In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I suppose I should take my own advice on this one, but sometimes I guess we all just can't resist. grin Tom Tom Thanks for your prompt and excellent response... Regards, Senad ___ --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] chan problem
Hello, the problem is nearly solved. When I restart the computer, Asterisk load prefectly but the ISDN calls doen't go. I must stop the Asterisk and run /etc/init.d/misdn-init start and then start Asterik. I have a Debian machine, I need to to do something like this: update-rc.d asterisk defaults update-rc.d misdn-init defaults but the problem is that Asterisk run before misdn-init, I and I want to start misd-init first. I dont know how to do. thanks a lot. 2007/6/21, [EMAIL PROTECTED] [EMAIL PROTECTED]: Thx, However it appears to be something else. Still need to find out what it is. Loading during boot does not work. After unloading (rmmod) modules mISDN, zaptel, wctdm etc, then reloading them manually in any particular order it works. On Mon, Jun 18, 2007 at 12:36:10PM -0400, Bob Chiodini wrote: [EMAIL PROTECTED] wrote: I experienced the same problem. The only way I could get both ISDN and analog working was unloading kernel modules for zaptel and mISDN after boot and then load them in the order: zaptel first and then mISDN. Still need to find out how to configure load order in linux. Hans, Have a look at the man page for modprobe.conf, specifically the install directive. There is an example of how to force the order. But it is already heavily abused. You may actually want to load one and not the other, and with that directive you can't . One alternative guess is the need to blacklist a third module. -- 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 ___ --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] chan problem
On Thu, Jun 21, 2007 at 10:05:47AM +0200, Josu Lazkano wrote: but the problem is that Asterisk run before misdn-init, I and I want to start misd-init first. I dont know how to do. # not sure if the first line is needed: update-rc.d -f remove asterisk update-rc.d defaults 30 10 -- 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] chan problem
On Thu, Jun 21, 2007 at 09:50:03AM +0200, [EMAIL PROTECTED] wrote: Thx, However it appears to be something else. Still need to find out what it is. Loading during boot does not work. After unloading (rmmod) modules mISDN, zaptel, wctdm etc, then reloading them manually in any particular order it works. Have you also unloaded the low-level misdn driver? One usual suspect whose name keeps popping up: hisax. TDM400P seems to use a certain chipset that is also used by some ISDN cards. So if you look at its aliases stirngs: alias: pci:vE159d0001svB1D9sd*bc*sc*i* alias: pci:vE159d0001svB118sd*bc*sc*i* alias: pci:vE159d0001svB119sd*bc*sc*i* alias: pci:vE159d0001svA9FDsd*bc*sc*i* alias: pci:vE159d0001svA8FDsd*bc*sc*i* alias: pci:vE159d0001svA800sd*bc*sc*i* alias: pci:vE159d0001svA801sd*bc*sc*i* alias: pci:vE159d0001svA908sd*bc*sc*i* alias: pci:vE159d0001svA901sd*bc*sc*i* Read: PCI cards with vendor ID E159, product ID 1 and a bunch of more specific sub-vendor IDs. The hisax driver has: alias: pci:vE159d0001sv*sd*bc*sc*i* That is: it is a generic driver that will try to probe all PCI cards with vendor ID E159 and product ID 1. I don't have misdn drivers installed, but you can narrow your search a bit by: grep e159 /lib/modules/`uname -r`/modules.pciids -- 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
[asterisk-users] mediant 2000 with asterik configuration
Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it - Get your own web address. Have a HUGE year through Yahoo! Small Business.___ --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] chan problem
update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't it? ___ --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 GUI
Troy Ayers wrote: I would have been convinced if you had not top-posted! heh Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob Ability to listen is a gift. People who have it apply data received into prosperity and greater good personally and collectively. Senad ___ --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] gtalk - no audio
I been to this scenario before. But I got mine working just last May 2007 and it appears to be stable now ready for some serious commercial application. Hint: if you have any experience with C, try to check with the source code related to the channels you are stressing down here. Well, it is not an easy task to do and not for the faint of heart. With a little luck and more of motivated creativity, you will get it working. Trust me, been there done that. Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage=Talk to me timeout=100 *gtalk.conf:* [general] context=default allowguest=yes bindaddr=172.25.123.18 [guest] disallow=all allow=ulaw context=gtalk This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. I first had similar problems, because I didn't use bindaddr in gtalk.conf. But that fixed it for me, but not for most other cases. Also, we all use the same network (same routing and NAT) and Gtalk version. Audio calls between regular Gtalk users is not a problem. This problem really puzzles me. Is it a channel gtalk problem, or do we need to look at other settings (network, client settings...)? I personnaly think we can rule out network config, since both successful and unsuccessful users work in the same lan. Is there anybody with experience in using channel gtalk? Should we start debugging? What can we learn from jabber debug logs? Any help is very much appreciated! koenvi ___ --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 config files and #include
On Thu, Jun 21, 2007 at 12:35:30PM +0530, Deepak Bhat wrote: Hi all, I am using asterisk version 1.2.18. I recently tried to change my asterisk configuration by using #include statements to include other config files in my extensions.conf and queues.conf files. My queues.conf is in /etc/asterisk. It includes several files which are in /etc/asterisk/queues. Each of these files contains the config of individual queues. Again each of the individual queue config files in /etc/asterisk/queues includes files which are in /etc/asterisk/queues/queue_members. The problem is that when I reload this config I get the following error: - *WARNING: Maaximum include level exceeded : 10* Has anyone encoutered this before and does anyone know what it means ?? Any help will be deeply appreciated as I have been unable to find any documentation on this. Sounds like a circular include: in extensions.conf: #include extensions.conf The circle may include more than one file. To trac this, enable debugging and debug logging. There is a debug comment for each included file. Unless you really have such a complex nesting structure of include files and want that constant changed. That it easy to do by a code change. I don't really see a reason to make this configurable, until someone shows me a case where this does not indicate a circular include. Hmmm... so should the error message be changed to: *WARNING: Maaximum include level exceeded : 10. Check for circular includes.* ? -- 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] chan problem
On Thu, Jun 21, 2007 at 10:41:29AM +0200, Josu Lazkano wrote: update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't it? Right. -- 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] mediant 2000 with asterik configuration
On Thu, Jun 21, 2007 at 01:40:39AM -0700, satish patel wrote: Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it I have no idea. But posting the same message under three different threads will not help. -- 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] chan problem
thank you Tzafrir. 2007/6/21, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, Jun 21, 2007 at 10:41:29AM +0200, Josu Lazkano wrote: update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't it? Right. -- 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 ___ --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] TDM400 one way calls
Dear All
I have a problem with a TDM400 card with 4 x FXS modules.
The card carries extensions only and there are no incoming lines.
I can make a call to the extension on this card with no problems.
However, when I try and call out I just get a busy signal.
I also get an error message (as shown at the bottom). Is this a problem?
Configs below:
[EMAIL PROTECTED] etc]# more zaptel.conf
fxoks=1-4
loadzone=uk
defaultzone=uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
;define trunks here
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;define channels
context=dialphone
signalling=fxo_ks
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity
usecallerid=yes
channel = 1-4
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
FAX1 = Zap/1
FAX2 = Zap/2
STREAMLINE1 = Zap/3
STREAMLINE2 = Zap/4
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;PRESS2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
;
[default]
;setupdial out
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
[dialphone]
exten = 601,1,Macro(oneline,${FAX1})
;
asterisk*CLI reload chan_zap.so
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
[Jun 21 10:24:26] WARNING[29786]: chan_zap.c:11072 process_zap: Ignoring
signalling
-- Reconfigured channel 1, FXO Kewlstart signalling
-- Reconfigured channel 2, FXO Kewlstart signalling
-- Reconfigured channel 3, FXO Kewlstart signalling
-- Reconfigured channel 4, FXO Kewlstart signalling
== Parsing '/etc/asterisk/users.conf': Found___
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[asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql .. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] TDM400 one way calls
Dear All
I have a problem with a TDM400 card with 4 x FXS modules.
The card carries extensions only and there are no incoming lines.
I can make a call to the extension on this card with no problems.
However, when I try and call a different extension I just get a busy signal.
I also get an error message (as shown at the bottom). Is this a problem?
Configs below:
[EMAIL PROTECTED] etc]# more zaptel.conf
fxoks=1-4
loadzone=uk
defaultzone=uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
;define trunks here
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;define channels
context=dialphone
signalling=fxo_ks
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity
usecallerid=yes
channel = 1-4
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
FAX1 = Zap/1
FAX2 = Zap/2
STREAMLINE1 = Zap/3
STREAMLINE2 = Zap/4
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;PRESS2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
;
[default]
;setupdial out
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
[dialphone]
exten = 601,1,Macro(oneline,${FAX1})
;
asterisk*CLI reload chan_zap.so
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
[Jun 21 10:24:26] WARNING[29786]: chan_zap.c:11072 process_zap: Ignoring
signalling
-- Reconfigured channel 1, FXO Kewlstart signalling
-- Reconfigured channel 2, FXO Kewlstart signalling
-- Reconfigured channel 3, FXO Kewlstart signalling
-- Reconfigured channel 4, FXO Kewlstart signalling
== Parsing '/etc/asterisk/users.conf': Found___
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Re: [asterisk-users] Asterisk config files and #include
Im sure its not a circular include. Like you said its mostly realted to the number of nested includes but the exact meaning is not clear to me. Anyways to get it working I have consolidated most of my queue config files and am not including anything from files that are included. Thanks! Tzafrir Cohen wrote: On Thu, Jun 21, 2007 at 12:35:30PM +0530, Deepak Bhat wrote: Hi all, I am using asterisk version 1.2.18. I recently tried to change my asterisk configuration by using #include statements to include other config files in my extensions.conf and queues.conf files. My queues.conf is in /etc/asterisk. It includes several files which are in /etc/asterisk/queues. Each of these files contains the config of individual queues. Again each of the individual queue config files in /etc/asterisk/queues includes files which are in /etc/asterisk/queues/queue_members. The problem is that when I reload this config I get the following error: - *WARNING: Maaximum include level exceeded : 10* Has anyone encoutered this before and does anyone know what it means ?? Any help will be deeply appreciated as I have been unable to find any documentation on this. Sounds like a circular include: in extensions.conf: #include extensions.conf The circle may include more than one file. To trac this, enable debugging and debug logging. There is a debug comment for each included file. Unless you really have such a complex nesting structure of include files and want that constant changed. That it easy to do by a code change. I don't really see a reason to make this configurable, until someone shows me a case where this does not indicate a circular include. Hmmm... so should the error message be changed to: *WARNING: Maaximum include level exceeded : 10. Check for circular includes.* ? ___ --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] chan problem
Hello, I have the same problem as the begining. I reinstall all the system and i have the same error: asterisk:/usr/src/asterisk-1.2.19# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel and wctdm modules are loaded correctly. And the zapata and zaptel files are correctly too. Thanks for all. 2007/6/21, Josu Lazkano [EMAIL PROTECTED]: thank you Tzafrir. 2007/6/21, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, Jun 21, 2007 at 10:41:29AM +0200, Josu Lazkano wrote: update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't it? Right. -- 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
No one faced a problem like this !! _ From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql .. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] gtalk - no audio
Would you be so kind to share your experience? I can read most of C language, but writing it is another thing. And I'm not familiar with the internals of Asterisk... Or maybe you could already confirm that my problem is related to NAT (client or Asterisk side, not sure) On 6/21/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I been to this scenario before. But I got mine working just last May 2007 and it appears to be stable now ready for some serious commercial application. Hint: if you have any experience with C, try to check with the source code related to the channels you are stressing down here. Well, it is not an easy task to do and not for the faint of heart. With a little luck and more of motivated creativity, you will get it working. Trust me, been there done that. Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage=Talk to me timeout=100 *gtalk.conf:* [general] context=default allowguest=yes bindaddr=172.25.123.18 [guest] disallow=all allow=ulaw context=gtalk This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. I first had similar problems, because I didn't use bindaddr in gtalk.conf. But that fixed it for me, but not for most other cases. Also, we all use the same network (same routing and NAT) and Gtalk version. Audio calls between regular Gtalk users is not a problem. This problem really puzzles me. Is it a channel gtalk problem, or do we need to look at other settings (network, client settings...)? I personnaly think we can rule out network config, since both successful and unsuccessful users work in the same lan. Is there anybody with experience in using channel gtalk? Should we start debugging? What can we learn from jabber debug logs? Any help is very much appreciated! koenvi ___ --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] gtalk - no audio
Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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 config files and #include
I don't have the source for 1.2.18 handy and didn't bother digging through my 1.4.4 tree looking but a quick grep for the exact error you see didn't reveal anything... although i greped the typo in maaximum However, correct that and that leads you to config.c #define MAX_INCLUDE_LEVEL 10 I suspect if your nesting a lot of includes you would probably need to up this level. I don't see a way to change this in asterisk.conf so I would suggest if you really need to go that deep in includes edit this option re-compile and be happy. NOTE *** This was in 1.4.4 maybe different in your version. I'm also not qualified to say from a quick glance if upping this limit has any negative impact but I would imagine it wouldn't and is more to help keep from causing loops. On Jun 21, 2007, at 6:37 AM, Deepak Bhat wrote: Im sure its not a circular include. Like you said its mostly realted to the number of nested includes but the exact meaning is not clear to me. Anyways to get it working I have consolidated most of my queue config files and am not including anything from files that are included. Thanks! Tzafrir Cohen wrote: On Thu, Jun 21, 2007 at 12:35:30PM +0530, Deepak Bhat wrote: Hi all, I am using asterisk version 1.2.18. I recently tried to change my asterisk configuration by using #include statements to include other config files in my extensions.conf and queues.conf files. My queues.conf is in /etc/asterisk. It includes several files which are in /etc/asterisk/queues. Each of these files contains the config of individual queues. Again each of the individual queue config files in /etc/asterisk/ queues includes files which are in /etc/asterisk/queues/queue_members. The problem is that when I reload this config I get the following error: - *WARNING: Maaximum include level exceeded : 10* Has anyone encoutered this before and does anyone know what it means ?? Any help will be deeply appreciated as I have been unable to find any documentation on this. Sounds like a circular include: in extensions.conf: #include extensions.conf The circle may include more than one file. To trac this, enable debugging and debug logging. There is a debug comment for each included file. Unless you really have such a complex nesting structure of include files and want that constant changed. That it easy to do by a code change. I don't really see a reason to make this configurable, until someone shows me a case where this does not indicate a circular include. Hmmm... so should the error message be changed to: *WARNING: Maaximum include level exceeded : 10. Check for circular includes.* ? ___ --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 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
I would first ask are do you have mysql client libraries installed Do you have them installed in the standard locations... I tend to never install anything in normal places for me it makes easier version control to put everything in specific places. did you try just running ./configure and watch for the part about mysql libraries did it find them? try just running make and see if the error gives you a bit more information about missing files. if you get around just that you can simply copy the .so file to your asterisk directory but ofcourse it's got to compile first. On Jun 21, 2007, at 5:52 AM, Khaled Chehab wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql …. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --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] chan problem
On Thu, Jun 21, 2007 at 12:48:46PM +0200, Josu Lazkano wrote: Hello, I have the same problem as the begining. I reinstall all the system and i have the same error: asterisk:/usr/src/asterisk-1.2.19# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel and wctdm modules are loaded correctly. cat /proc/zaptel/* And the zapata and zaptel files are correctly too. cat /etc/zaptel.conf BTW: what is the output of: ./xpp/utils/genzaptelconf -l -- 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] asterisk 1.4.1 app_addon_sql_mysql
Do you have MySQL installed in your machine??? On 6/21/07, Khaled Chehab [EMAIL PROTECTED] wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql …. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 config files and #include
On Thu, Jun 21, 2007 at 04:07:03PM +0530, Deepak Bhat wrote: Im sure its not a circular include. Like you said its mostly realted to the number of nested includes but the exact meaning is not clear to me. I repeat: To trace this, enable debugging and debug logging. There is a debug comment for each included file. enable 'debug' for some log file in logger.conf , and then run: logger reload reload -- 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] chan problem
Hello, I have solved. I must delete the netjetpci module from /etc/modprobe.d/blacklist: blacklist netjetpci thanks for all 2007/6/21, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, Jun 21, 2007 at 12:48:46PM +0200, Josu Lazkano wrote: Hello, I have the same problem as the begining. I reinstall all the system and i have the same error: asterisk:/usr/src/asterisk-1.2.19# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel and wctdm modules are loaded correctly. cat /proc/zaptel/* And the zapata and zaptel files are correctly too. cat /etc/zaptel.conf BTW: what is the output of: ./xpp/utils/genzaptelconf -l -- 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 ___ --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] Problem with Remote-Hold/MusicOnHold
Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends Remote hold message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to extension 20. But Asterisk doesn't send Remote retrieve, so the mosic-on-hold of the carrier doesn't stop. You can hear the person on Mobile A, but the person on Mobile A only gets music. Anybody with the same issue? It happens on a bristuffed Asterisk 1.2.19. Is there a way to stop sending Remote hold or a way to send Remote retrieve? See here the the cli output with pri debug span 4 at starting attended transfer: 4 Protocol Discriminator: Q.931 (8) len=7 4 Call Ref: len= 1 (reference 214/0xD6) (Terminator) 4 Message type: NOTIFY (110) 4 [27 01 f9] 4 Notification indicator (len= 3): Ext: 1 Remote hold (121) -- Started music on hold, class 'default', on channel 'Zap/10-1' -- Playing 'pbx-transfer' (language 'de') Best regards, Gunnar Schaller ___ --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] ChanSkype
Hello, I recently installed chanskype on my asterisk box and it works like a dream, can phone out. But no idea how to setup the incoming calls, every time I phone my skype name it just connects and disconnect the call right away. I get the following on asterisk -rvv Verbosity was 1 and is now 14 == Sent cmd 'GET CALL 175 TYPE' to fd 18 on Skype dev 'skype1' == Sent cmd 'GET CALL 175 PARTNER_HANDLE' to fd 18 on Skype dev 'skype1' == Sent cmd 'GET CALL 175 PSTN_NUMBER' to fd 18 on Skype dev 'skype1' == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1' == Sent cmd 'ALTER CALL 175 END HANGUP' to fd 18 on Skype dev 'skype1' == Unknown event 'ALTER CALL 175 END HANGUP' from Skype device 'skype1' == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1' Any one got some advice ? Kind Regards, Kyle Vorster ___ --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 Faxing
Any one know more about this, Please assist if possible. Kyle Vorster wrote: Any one able to assist, Please Paradise Dove wrote: so how to avoid CPC?? On 6/14/07, C F [EMAIL PROTECTED] wrote: Its called CPC On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote: Hello, Sorry if this is a real dumb question but when sending a fax and the end user does not enable fax on their side and then just hangs up does not force asterisk to end the call. So it keeps the trunk open until its killed by a Flash Operator. Please assist if any one understands me. Kind Regards, Kyle Virster ___ --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
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
Yes mysql installed [EMAIL PROTECTED] asterisk-1.4.5]# rpm -q mysql mysql-4.1.20-2.RHEL4.1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leonardo Kamache (Gmail) Sent: Thursday, June 21, 2007 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql Do you have MySQL installed in your machine??? On 6/21/07, Khaled Chehab [EMAIL PROTECTED] wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql .. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Using Queue - Zap problems (PRI)
I have asterisk 1.4 using Queue application. I have this error I must restart Asterisk to correct it. Any Ideas?? log: [Jun 20 16:42:27] WARNING[29339] channel.c: Unexpected control subclass '17' [Jun 20 16:42:38] NOTICE[29337] app_queue.c: No one is answering queue 'myqueue' (13/12/0) [Jun 20 16:44:16] WARNING[30203] file.c: Failed to write frame [Jun 20 16:45:11] WARNING[8044] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. [Jun 20 16:45:20] WARNING[8044] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. [Jun 20 16:45:27] WARNING[8044] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. [Jun 20 16:45:31] WARNING[8044] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
Do you have the mysql client and header files installed? On Thu, 2007-06-21 at 04:11 -0700, Khaled Chehab wrote: Yes mysql installed [EMAIL PROTECTED] asterisk-1.4.5]# rpm -q mysql mysql-4.1.20-2.RHEL4.1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leonardo Kamache (Gmail) Sent: Thursday, June 21, 2007 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql Do you have MySQL installed in your machine??? On 6/21/07, Khaled Chehab [EMAIL PROTECTED] wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql .. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 1.4.1 app_addon_sql_mysql
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Thursday, June 21, 2007 7:12 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql Yes mysql installed [EMAIL PROTECTED] asterisk-1.4.5]# rpm -q mysql mysql-4.1.20-2.RHEL4.1 You need mysql-devel - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] AudioCodes Gateway and Asterisk
Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 Unsupported Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid___ --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] ChanSkype
On 6/21/07, Kyle Vorster [EMAIL PROTECTED] wrote: Hello, I recently installed chanskype on my asterisk box and it works like a dream, can phone out. But no idea how to setup the incoming calls, every time I phone my skype name it just connects and disconnect the call right away. ... Any one got some advice ? My advice: contact the developer of ChanSkype. You have to pay for that, right? Hopefully, it comes with some support. In the mean time, make sure your incoming call's context exists, ensure that you have an s and i extension in that extension just in case the number comes in differently than how you expect, and put some no-ops in, maybe have it echo the EXTEN variable. You know, basic troubleshooting. Good luck, David ___ --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] retreiving callid of call from the dial application
Hi, I am making calls from the dial plan using the dial application. Due to technical requirements I need to find out the sip call-id used in the dialog initiated by the dial application. I dont see any straight forward way of doing this so I am looking for answers. There is a sip callid session variable but the problems is that dial is a blocking call and the dialog ends when dial returns. I saw a similar post on the users list but there was no apparent solution suggested. Our setup permits simultaneous calls as well so I need to retreive call id's for each call made. I would appreciate any suggestions, be that workarounds or code hacks. Regards, Danish ps: I apologize for cross posting, but I am not sure which forums was better suited for this question. ___ --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] Zaptel wct2xxp driver causes LEDs to go black and RED alarm
Hi there, I've got two Asterisk hosted PBX servers with Digium TE210P cards attached by a E1 cable to Port 1 on each. On startup, both cards flash red, alternating between ports 1 and 2. When server #1 loads the Zaptel module and drivers, Port 1 status LED goes green. When server #2 loads the same module and drivers, Port 1 status LED goes completely blank. Unloading the wct2xxp module causes the flashing red LEDs to come back. I've tried swapping cable ends and cards between the two machines, but the problem LED always stays with server #2. So, I think there is something misconfigured with server #2, but the configuration file on both servers is identical. zaptel.conf: loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Any clues what could cause a GREEN alarm on one end with a RED alarm on the other and no LED light as soon as the wct2xxp driver is loaded? Thanks for the help, Jason Carter DLS Internet Services ___ --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] ENC: Action Originate (Asterisk Manager) X Monitor()
Hi people! I need a help.
I connect with Asterisk Manager and execute an Originate Action that asks
for Asterisk to call for a number 09194 and to transfer to peer 101.
This command enters in my dial plan extension that I made exclusively for
tests.
The problem is:
If I do the call through ORIGINATE Action, the Monitor() just makes the
record file, but this is empty, if I do the call to SoftPhone the record
file is made normally.
My Action Originate:
Action: Originate
ActionID: 1BV2bwdI_#Ps20070620175008
Channel: Local/09194
Exten: 101
Context: ramais
Priority: 1
Variable: ACTIONID=1BV2bwdI_#Ps20070620175008
Async: True
Part of my dial plan:
exten = _09194,1,set(SCREEN_FILE=/calls/records/TESTE_${RAND(1,9)}
exten = _09194,2,Monitor(wav,${SCREEN_FILE},m)
exten = _09194,3,Dial(SIP/TmaisMG/9194,40,rtw)
exten = _091945,4,HangUp()
I checked, with SIP SET DEBUG, if it would be CODECs problems, all calls are
done by 'ulaw' CODEC.
My Channel, my peer and my SoftPhone are configured to use just 'ulaw'
CODEC.
I'm using Asterisk 1.4.2.
Regards!
Cordiality,
Moacir O. de Souza Junior
Belo Horizonte - Minas Gerais - Brasil
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[asterisk-users] identifying what a user pressed to reach my phone
I am a new trixbox user. One of the things I'd like to get working is being able to tell if a user is calling me by directly dialing my extension, or if they pressed 1 for sales. When they press 1, it rings a group of phones, and the call is almost always for someone else. So I'm always looking at my phone when it rings, trying to recognize the incoming number and decide if I should answer it. My idea was to setup line 2 on the phones as another extension. So each user would have 101 or 102, etc. as their regular extension. Each users line 2 would be configured as extension 201, 202, etc. Then I would have the 'press 1 for sales' function ring the 201, 202 group of phones. You'd know how the user reached you by seeing if line 1 or line 2 was lighting up during an incoming call. Is this a good way to do it? Also, how do I even begin to setup this automated attendant menu in Trixbox? (where I'll ask users to dial the party's extension or press 1 for sales, etc.) I've dug around in the menus but I don't see anything resembling this. Thanks, -Ryan ___ --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] AudioCodes Gateway and Asterisk
On 6/21/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 Unsupported Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Are you sure, your asterisk supports G729? It isn't supported by default, you need additional modules or hardware cards for G729 support. If it is - what are you using for G729 - that might help to identify the problem. Regards, Atis ___ --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] identifying what a user pressed to reach my phone
The IVR is your auto attendant. Look in the modules to install it if you haven't already As for identifying the calll with some custom programming, you could tweak the callerid. Check the taug.Cain a day or so for a script I tweaked which would be a good start on cid editing D Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Thu Jun 21 10:51:42 2007 Subject: [asterisk-users] identifying what a user pressed to reach my phone I am a new trixbox user. One of the things I'd like to get working is being able to tell if a user is calling me by directly dialing my extension, or if they pressed 1 for sales. When they press 1, it rings a group of phones, and the call is almost always for someone else. So I'm always looking at my phone when it rings, trying to recognize the incoming number and decide if I should answer it. My idea was to setup line 2 on the phones as another extension. So each user would have 101 or 102, etc. as their regular extension. Each users line 2 would be configured as extension 201, 202, etc. Then I would have the 'press 1 for sales' function ring the 201, 202 group of phones. You'd know how the user reached you by seeing if line 1 or line 2 was lighting up during an incoming call. Is this a good way to do it? Also, how do I even begin to setup this automated attendant menu in Trixbox? (where I'll ask users to dial the party's extension or press 1 for sales, etc.) I've dug around in the menus but I don't see anything resembling this. Thanks, -Ryan ___ --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] gtalk - no audio
what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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] identifying what a user pressed to reach my phone
Get a sheet of paper, write down your menu structure find the screen where you can record to asterisk record your voice menu based on what you wrote down then go to the ivr menu create a new one that matched what you wrote down choose the recording that you recorded earlier as the sound that plays for this menu. Your ring group can add something to the caller id take a look at the ring group setting screen. Ryan Stille wrote: I am a new trixbox user. One of the things I'd like to get working is being able to tell if a user is calling me by directly dialing my extension, or if they pressed 1 for sales. When they press 1, it rings a group of phones, and the call is almost always for someone else. So I'm always looking at my phone when it rings, trying to recognize the incoming number and decide if I should answer it. My idea was to setup line 2 on the phones as another extension. So each user would have 101 or 102, etc. as their regular extension. Each users line 2 would be configured as extension 201, 202, etc. Then I would have the 'press 1 for sales' function ring the 201, 202 group of phones. You'd know how the user reached you by seeing if line 1 or line 2 was lighting up during an incoming call. Is this a good way to do it? Also, how do I even begin to setup this automated attendant menu in Trixbox? (where I'll ask users to dial the party's extension or press 1 for sales, etc.) I've dug around in the menus but I don't see anything resembling this. Thanks, -Ryan ___ --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] Improving Asterisk's DNS support
On 6/20/07, Steven [EMAIL PROTECTED] wrote: I could understand if it couldn't register to an ITSP or similar. But, (I had this happen today) asterisk takes forever to start up and SIP phones can not register to it. DNS should not need to be used for anything in asterisk except registering to VOIP providers and maybe external SQL from the dialplan. If there are reverse lookups being done, I do not see the output of it. Steven, If you are using a hostname for an ITSP and DNS fails, it will take FOREVER for the SIP channel driver to load/reload/do anything that requires a DNS lookup. This will in some cases block the rest of Asterisk but will certainly make anything that depends on SIP break - until the DNS request finally fails. I have started a new thread on -dev about this... -- Kristian Kielhofner ___ --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] TDM800P - zaptel service startup problem
Dear Team, I have installed digium TDM800P card. its include 1 quad fxo module and 2 FXO modules. I installed zaptel 1.2.18, libpri-1.2.4 and asterisk 1.2.19. I installed all zaptel drivers , asterisk without any problem. following are my /etc/zapel.conf settings fxsks=1,2,3,4,5,6 /etc/sysconfig/zaptel MODULES=$MODULES wctdm8xxp The proble is once i reboot the server, zaptel service failed. error message is : Loading zaptel framework: [ OK ] Waiting for zap to come online...Error: missing /dev/zap! Please give me a feedback on this regard. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --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] Zaptel wct2xxp driver causes LEDs to go black and RED alarm
Have you checked to ensure the card in server #2 is jumpered for E1? On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wrote: Hi there, I've got two Asterisk hosted PBX servers with Digium TE210P cards attached by a E1 cable to Port 1 on each. On startup, both cards flash red, alternating between ports 1 and 2. When server #1 loads the Zaptel module and drivers, Port 1 status LED goes green. When server #2 loads the same module and drivers, Port 1 status LED goes completely blank. Unloading the wct2xxp module causes the flashing red LEDs to come back. I've tried swapping cable ends and cards between the two machines, but the problem LED always stays with server #2. So, I think there is something misconfigured with server #2, but the configuration file on both servers is identical. zaptel.conf: loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Any clues what could cause a GREEN alarm on one end with a RED alarm on the other and no LED light as soon as the wct2xxp driver is loaded? Thanks for the help, Jason Carter DLS Internet Services ___ --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] AudioCodes Gateway and Asterisk
On 6/21/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 Unsupported Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Are you sure, your asterisk supports G729? It isn't supported by default, you need additional modules or hardware cards for G729 support. If it is - what are you using for G729 - that might help to identify the problem. Regards, Atis If the AudioCodes is sending back that 415, the Message Log in the AudioCodes is invaluable. Set your debug level to 5/6 and watch it while you make test calls. Once you learn how to interpret this output, you'll be well on your way with AudioCodes. If G729 is active on the MP, but still giving back that error, G729 might not be in a profile if you are using them. Also, firmware that comes on the MPs is normally sorta buggy, ask your reseller for the latest version. http://www.abptech.com/support/faqs/ Regards, Shanon ABP Technology ___ --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] Forward to my phones the domain of the CALLERID in incoming URI calls
Is there a way I can forward to my phones the domain of the CALLERID in the CALLERID(number) field of INVITE messages, when some call arrives to my Asterisk? What happens in my architecture is this: INVITE [EMAIL PROTECTED] INVITE [EMAIL PROTECTED]'s_IP --- Asterisk --- john's_phone From: Mary sip:[EMAIL PROTECTED]From: Mary sip:[EMAIL PROTECTED]'s_IP As shown, Asterisk substitutes the domain of the caller contact in the From field of INVITE messages that are sent to the destination phone by Asterisk's IP address. That way, our phones just display Mary and mary when I want them to display Mary and [EMAIL PROTECTED], so that john can be aware that Mary is from an outside domain. Any ideas? How should be my extensions.conf so this can be possible? Regards, Ricardo. ___ --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] Bug in Ex-Girlfriend logic?
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/5000-0a281f80, ) in new
stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/5000-0a281f80, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/5000-0a281f80, 19256002182)
in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(SIP/5000-0a281f80,
tt-monkeys) in new stack
-- SIP/5000-0a281f80 Playing 'tt-monkeys' (language 'en')
However, when I change the extension match to:
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the
caller id number is 19256002182 from the NoOp() when it does work.
This had me stumped for a while, until I realized that the following
_DOES_ work:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,NoOp(Foo)
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
Yes. That's right. In order for the ex-girlfriend logic to match a
caller id of 19256002182 against 5000, the same context also needs to
have an extension for 5000, even if you intend to do nothing with it.
I'd never noticed this before, because normally you'd provision the 5000
extension FIRST and then the 5000/19256002182 after that.
Seems like a bug to me Problem was reproduced in 1.2.13, 1.2.19 and
1.4.4.
Doug.
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Re: [asterisk-users] gtalk - no audio
I haven't changed rtp.conf from original installation. So the values are: rtpstart=1 rtpend=2 I should maybe give it a try with a lower rtpstart. What do you mean by turning on NAT? Are you referring to parameter bindaddr in gtalk.conf? (found that on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk) Thanks already! On 6/21/07, Joseph Bajin [EMAIL PROTECTED] wrote: what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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] Zaptel wct2xxp driver causes LEDs to go black and RED alarm
Yes, both cards are jumpered for E1. Any other ideas? Jason :) James Texter wrote: Have you checked to ensure the card in server #2 is jumpered for E1? On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wrote: Hi there, I've got two Asterisk hosted PBX servers with Digium TE210P cards attached by a E1 cable to Port 1 on each. On startup, both cards flash red, alternating between ports 1 and 2. When server #1 loads the Zaptel module and drivers, Port 1 status LED goes green. When server #2 loads the same module and drivers, Port 1 status LED goes completely blank. Unloading the wct2xxp module causes the flashing red LEDs to come back. I've tried swapping cable ends and cards between the two machines, but the problem LED always stays with server #2. So, I think there is something misconfigured with server #2, but the configuration file on both servers is identical. zaptel.conf: loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Any clues what could cause a GREEN alarm on one end with a RED alarm on the other and no LED light as soon as the wct2xxp driver is loaded? Thanks for the help, Jason Carter DLS Internet Services ___ --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] Improving Asterisk's DNS support
Kristian Kielhofner wrote: On 6/20/07, Steven [EMAIL PROTECTED] wrote: I could understand if it couldn't register to an ITSP or similar. But, (I had this happen today) asterisk takes forever to start up and SIP phones can not register to it. DNS should not need to be used for anything in asterisk except registering to VOIP providers and maybe external SQL from the dialplan. If there are reverse lookups being done, I do not see the output of it. Steven, If you are using a hostname for an ITSP and DNS fails, it will take FOREVER for the SIP channel driver to load/reload/do anything that requires a DNS lookup. This will in some cases block the rest of Asterisk but will certainly make anything that depends on SIP break - until the DNS request finally fails. I have started a new thread on -dev about this... I experienced this exact problem last night on my personal box. My sip provider went un-reachable (Teliax requires the use of hostnames). When that happened, I couldn't even call my local phone extensions. Everything SIP was locked hard until it finally timed out. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
[asterisk-users] CDR
I am using asterisk 1.4.5 with asterisk-addons-1.4.2 On /var/log/asterisk/cdr-csv/Master.csvthe unique id But in mysql database ,the unique id is not shown ,how can I fix it .. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] CDR
I am using asterisk 1.4.5 with asterisk-addons-1.4.2 On /var/log/asterisk/cdr-csv/Master.csvthe unique id showed But in mysql database ,the unique id is not shown ,how can I fix it .. Regards _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 GUI
On Jun 20, 2007, at 5:04 PM, Troy Ayers wrote: I would have been convinced if you had not top-posted! heh Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob OK, Now I'm confused... I was prepared to accept Rob's argument due its beautiful, flawless logic. But Troy has a valid point: Rob did top-post, invalidating his point. But so did Troy, invalidating his point, so now I'm stuck. Whatever shall I do? I think I'll just stick with my own opinion, seeing as both Rob and Troy are obviously idiots. (duh!) ;-) Tom ___ --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] TDM800P - zaptel service startup problem
The TDM800P uses this driver MODULES=MODULES wctdm24xxp Maybe you can try with it. -- Ing. Arturo Ochoa N Network Administrator Electrosystems, Vidura Senadeera wrote: Dear Team, I have installed digium TDM800P card. its include 1 quad fxo module and 2 FXO modules. I installed zaptel 1.2.18, libpri-1.2.4 and asterisk 1.2.19. I installed all zaptel drivers , asterisk without any problem. following are my /etc/zapel.conf settings fxsks=1,2,3,4,5,6 /etc/sysconfig/zaptel MODULES=$MODULES wctdm8xxp The proble is once i reboot the server, zaptel service failed. error message is : Loading zaptel framework: [ OK ] Waiting for zap to come online...Error: missing /dev/zap! Please give me a feedback on this regard. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk http://www.debug.lk ___ --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] CDR
On Jun 21, 2007, at 10:33 AM, Khaled Chehab wrote: I am using asterisk 1.4.5 with asterisk-addons-1.4.2 On /var/log/asterisk/cdr-csv/Master.csvthe unique id But in mysql database ,the unique id is not shown ,how can I fix it .. did you see http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? about changing the compile time option -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --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] gtalk - no audio
When either someone inside asterisk or gtalk makes the call, one could see the randomness of the ports being used by the RTP. Btw, what does jingle.conf and gtalk.conf have in common? In my experience, the two of them should go hand in hand as the channel keeps looking into it. I have a success with the googletalk to asterisk and vice versa either all of them inside the NAT firewall or one of them is outside the NAT firewall. What part of the channel code is responsible for the handling of calls like the ringing etcetera? I haven't changed rtp.conf from original installation. So the values are: rtpstart=1 rtpend=2 I should maybe give it a try with a lower rtpstart. What do you mean by turning on NAT? Are you referring to parameter bindaddr in gtalk.conf? (found that on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk) Thanks already! On 6/21/07, Joseph Bajin [EMAIL PROTECTED] wrote: what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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] gtalk - no audio
it does not appear that the RTP port could be higher or lower in a particular instance of a call. Philippe, what part of the channel code handles the ringing and dialling. From my experience here, making a call from googletalk to a voip phone inside a firewalled environment does not pose any problem. But making call from voip phone to googletalk is kinda tricky. what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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] gtalk - no audio
Philippe, what part of the channel code handles the ringing and dialling.
From my
experience here, making a call from googletalk to a voip phone inside a
firewalled
environment does not pose any problem. But making call from voip phone to
googletalk is
kinda tricky.
Well, chan_gtalk being a channel, its PBX functions are all gathered
in a ast_channel_tech structure :
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
.type = Gtalk,
.description = Gtalk Channel Driver,
.capabilities = ((AST_FORMAT_MAX_AUDIO 1) - 1),
.requester = gtalk_request,
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
.bridge = ast_rtp_bridge,
.call = gtalk_call,
.hangup = gtalk_hangup,
.answer = gtalk_answer,
.read = gtalk_read,
.write = gtalk_write,
.exception = gtalk_read,
.indicate = gtalk_indicate,
.fixup = gtalk_fixup,
.send_html = gtalk_sendhtml,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
demuel, do you have an extensions.conf (or ael) snippet for a VoIP
phone - Asterisk - GoogleTalk call scenario? I wonder why this does
not work in your case.
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Re: [asterisk-users] ATT: Brian Fertig
You might want to call: 302.338.9601 On 6/20/07, Dean Collins [EMAIL PROTECTED] wrote: Hi Brian, Trying to get in touch, please call or email Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --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] TDM400 one way calls
What phone are you trying to dial? (ZAP/1?)
What is the CLI output when you dial?
What number are you dialing? 601?
On 6/21/07, Matt Scott [EMAIL PROTECTED] wrote:
Dear All
I have a problem with a TDM400 card with 4 x FXS modules.
The card carries extensions only and there are no incoming lines.
I can make a call to the extension on this card with no problems.
However, when I try and call a different extension I just get a busy signal.
I also get an error message (as shown at the bottom). Is this a problem?
Configs below:
[EMAIL PROTECTED] etc]# more zaptel.conf
fxoks=1-4
loadzone=uk
defaultzone=uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
;define trunks here
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;define channels
context=dialphone
signalling=fxo_ks
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity
usecallerid=yes
channel = 1-4
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
FAX1 = Zap/1
FAX2 = Zap/2
STREAMLINE1 = Zap/3
STREAMLINE2 = Zap/4
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;PRESS2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
;
[default]
;setupdial out
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
[dialphone]
exten = 601,1,Macro(oneline,${FAX1})
;
asterisk*CLI reload chan_zap.so
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
[Jun 21 10:24:26] WARNING[29786]: chan_zap.c:11072 process_zap: Ignoring
signalling
-- Reconfigured channel 1, FXO Kewlstart signalling
-- Reconfigured channel 2, FXO Kewlstart signalling
-- Reconfigured channel 3, FXO Kewlstart signalling
-- Reconfigured channel 4, FXO Kewlstart signalling
== Parsing '/etc/asterisk/users.conf': Found
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Re: [asterisk-users] gtalk - no audio
If you are behind a firewall, you may need to turn on NAT in order for the RTP to be able to connect to each other. If you have wireshark or able to get a TCPDump, make the call that fails and look at the media anchors. For me (when I had the exact same problem), Gtalk came in with a media port of like 5800 or something in that range. I was only looking at 1 and above. So of course, I didn't get bi-directional audio. Once I changed that rtpstart to 2000, I was able to get things working again. Plus I had to turn on NAT support. On 6/21/07, Koen Van Impe [EMAIL PROTECTED] wrote: I haven't changed rtp.conf from original installation. So the values are: rtpstart=1 rtpend=2 I should maybe give it a try with a lower rtpstart. What do you mean by turning on NAT? Are you referring to parameter bindaddr in gtalk.conf? (found that on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk) Thanks already! On 6/21/07, Joseph Bajin [EMAIL PROTECTED] wrote: what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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
[asterisk-users] Use of ChanSpy
How can I use the Asterisk command ChanSpy If I need to spy on a call? I already added the function to the extensions.conf, and I get the beeps, but then what do I do??? I don't understand the use of this function. Best 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
Re: [asterisk-users] Use of ChanSpy
Carlos, From what I understand, the premise is that if you dial the ChanSpy extension from another phone, it should place you in a position to listen in on a bridged call (a call whose media runs 'through' Asterisk). -- Alex On Thu, 21 Jun 2007, Carlos Garcia Mujica wrote: How can I use the Asterisk command ChanSpy If I need to spy on a call? I already added the function to the extensions.conf, and I get the beeps, but then what do I do??? I don't understand the use of this function. Best Regards -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Console channels with two sound cards?
If asterisk is running on a system that has two sound cards, is it possible to run two Console channels? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Polycoms lose registration and won't re-register
Maybe you could do a test with : a; using the latest polycom administration guide (examples found on voip-info.org) to supply configs and firmware b; use latest firmware (2.1.0.something) c; if the issue doesn't go away, try ethereal between a 'misbehaving' phone and the switch to see what the phone is actually sending out.. (in the mean time use sip debug on the asterisk CLI) cheers.. ___ --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] identifying what a user pressed to reach my phone
RS == Ryan Stille [EMAIL PROTECTED] writes: RS I am a new trixbox user. One of the things I'd like to get working RS is being able to tell if a user is calling me by directly dialing RS my extension, or if they pressed 1 for sales. When they press 1, RS it rings a group of phones, and the call is almost always for RS someone else. So I'm always looking at my phone when it rings, RS trying to recognize the incoming number and decide if I should RS answer it. My favourite solutions to this one is to either change the ringing sound for the direct calls (this is phone specific; Snoms can do it at least) or to add something to CALLERID(name). /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: [asterisk-users] gtalk - no audio
Yeah, just the same as the sample configuration by mog. However, if I am using
a gtalk
application in asterisk to dial googletalk buddy, my voip phone is suddenly
connected to
the googletalk buddy though at the googletalk client software it is still
waiting to be
accepted or not accepted. I mean from my voip phone perspective, there is just
one ring
to make a call to the googletalk buddy unlike in the jingle application wherein
there
are successive ring before the googletalk buddy accepts the call.
please let me know if this is not clear to you and thanks a lot.
Philippe, what part of the channel code handles the ringing and dialling.
From my
experience here, making a call from googletalk to a voip phone inside a
firewalled
environment does not pose any problem. But making call from voip phone to
googletalk
is
kinda tricky.
Well, chan_gtalk being a channel, its PBX functions are all gathered
in a ast_channel_tech structure :
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
.type = Gtalk,
.description = Gtalk Channel Driver,
.capabilities = ((AST_FORMAT_MAX_AUDIO 1) - 1),
.requester = gtalk_request,
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
.bridge = ast_rtp_bridge,
.call = gtalk_call,
.hangup = gtalk_hangup,
.answer = gtalk_answer,
.read = gtalk_read,
.write = gtalk_write,
.exception = gtalk_read,
.indicate = gtalk_indicate,
.fixup = gtalk_fixup,
.send_html = gtalk_sendhtml,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
demuel, do you have an extensions.conf (or ael) snippet for a VoIP
phone - Asterisk - GoogleTalk call scenario? I wonder why this does
not work in your case.
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Re: [asterisk-users] gtalk - no audio
turning NAT won't give assurance that you get an audio. remember, there are two protocols involve here. one is sip and the other one is rtp. the rtp protocol is quite serious to deal with specially if the ports it uses are kinda random and one has to exert a lot of effort to configure the firewall to allows ranges of ports. I got both audio both ways. from googletalk buddy to voip phone behind a firewalled asterisk and vice versa. If you are behind a firewall, you may need to turn on NAT in order for the RTP to be able to connect to each other. If you have wireshark or able to get a TCPDump, make the call that fails and look at the media anchors. For me (when I had the exact same problem), Gtalk came in with a media port of like 5800 or something in that range. I was only looking at 1 and above. So of course, I didn't get bi-directional audio. Once I changed that rtpstart to 2000, I was able to get things working again. Plus I had to turn on NAT support. On 6/21/07, Koen Van Impe [EMAIL PROTECTED] wrote: I haven't changed rtp.conf from original installation. So the values are: rtpstart=1 rtpend=2 I should maybe give it a try with a lower rtpstart. What do you mean by turning on NAT? Are you referring to parameter bindaddr in gtalk.conf? (found that on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk) Thanks already! On 6/21/07, Joseph Bajin [EMAIL PROTECTED] wrote: what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --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
Re: [asterisk-users] different codec for different extensions
Configure the channels with the proper disallow= and allow= lines, and Asterisk should figure the rest out. I could be making drastic assumptions about your situation, but it seems like this: -sip.conf [userX] ... context=internal disallow=all allow=gsm allow=ulaw ... [fax] ... disallow=all allow=ulaw ... Then any IVRs that userX accesses should be in gsm because it's the preferred codec? Assuming that the gsm sound files ARE installed? You might experiment with this. But when userX is bridged to the fax channel, ulaw is the only one the fax channel allows, so it's chosen on both ends. Shouldn't this work? Mojo Nasir Iqbal wrote: Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --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] Blind xfer issue -- URGENT!
Use the dialplan show CLI command (show dialplan in 1.2) to show
you exactly what asterisk has picked up, and scan it for aforementioned
leaks.
Rizwan Hisham wrote:
Then i think u should use Atis's idea of using transfer_context
variable...you should set it inside your dialplan and it should be
the first thing you do in your dialplan.
Are you sure there is no leak in your dialplan, because asterisk cant
transfer your caller to an extension it cant find. There must be leak,
check if you are using any wrong extension patterns like _XXX. or
something like that.
On 6/19/07, *Jay Moore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The way I have my dialplan set up, the callers shouldn't be able to make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include callcenter/extension_ans.conf
include = answering-service
Which includes my callcenter answering service extensions conf file and
includes the 'answering-service' context:
{callcenter/extension_ans.conf}
[answering-service]
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
Which then parses all incoming calls (you can see the rest of the
dialplan in my previous message).
I'm not sure what I'm doing wrong. It seems to me I'm doing everything
properly. Callers should not be able to transfer (no 'T' in the Queue()
command), and they should not be able to dial any extension.
I'm completely lost here.
Jay
Rizwan Hisham wrote:
I dont know how to solve your transfer problem, but i have an
idea which
you
can use to overcome this abnormality.
You should restrict the callers with context which includes only
your local
office extensions.
I assume all your incoming calls fall in [default] context. what
you should
do is:
[default]
include= localext
exten= _X.,1,Noop(Incoming call received)
[localext]
*This context should include all your office extensions**
This way, caller can only transfer himself within your office
extensions.
I hope you get my point
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Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com http://www.axvoice.com
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[asterisk-users] Looking to buy VoIP or Hosting Company
We're looking to buy a VoIP company that does wholesale orig and term services. Ideally, $90,000 a month or more in revenue but will look at lower volumes. Also, a webhosting company. Geographic location can be anywhere in USA or Canada. Send email with the word CONFIDENTIAL in subject line. Send a little general info about what you have and what you have in mind. If it looks interesting, we'll sign an NDA and go from there. Reply to: Gregory [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hotline with Polycom
Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0:number). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom? The idea is that you pick up the handset and it automatically dials a number. It will be used in a foyer or front door. Many thanks David. ___ --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] Blind xfer issue -- URGENT!
It sounds more like the agents are making the transfers...
If a caller were to transfer a call (#0 1555-555-1212), they would be
transferring the AGENT to the that number, not themselves!
Either way, the caller SHOULD be disconnected after the transfer. (Or
perhaps leaked somewhere else into the dialplan they shouldn't be going,
which lets them dial out long-distance.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, June 21, 2007 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind xfer issue -- URGENT!
Use the dialplan show CLI command (show dialplan in 1.2) to show
you exactly what asterisk has picked up, and scan it for aforementioned
leaks.
Rizwan Hisham wrote:
Then i think u should use Atis's idea of using transfer_context
variable...you should set it inside your dialplan and it should be
the first thing you do in your dialplan.
Are you sure there is no leak in your dialplan, because asterisk cant
transfer your caller to an extension it cant find. There must be leak,
check if you are using any wrong extension patterns like _XXX. or
something like that.
On 6/19/07, *Jay Moore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The way I have my dialplan set up, the callers shouldn't be able to
make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include callcenter/extension_ans.conf
include = answering-service
Which includes my callcenter answering service extensions conf file
and
includes the 'answering-service' context:
{callcenter/extension_ans.conf}
[answering-service]
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
Which then parses all incoming calls (you can see the rest of the
dialplan in my previous message).
I'm not sure what I'm doing wrong. It seems to me I'm doing
everything
properly. Callers should not be able to transfer (no 'T' in the
Queue()
command), and they should not be able to dial any extension.
I'm completely lost here.
Jay
Rizwan Hisham wrote:
I dont know how to solve your transfer problem, but i have an
idea which
you
can use to overcome this abnormality.
You should restrict the callers with context which includes only
your local
office extensions.
I assume all your incoming calls fall in [default] context. what
you should
do is:
[default]
include= localext
exten= _X.,1,Noop(Incoming call received)
[localext]
*This context should include all your office extensions**
This way, caller can only transfer himself within your office
extensions.
I hope you get my point
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com http://www.axvoice.com
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Re: [asterisk-users] hotline with Polycom
We use this hotline / auto-dial functionality in our Polycoms. In the phone-specific XML config file we have the following entry: call autoOffHook call.autoOffHook.1.enabled=1 call.autoOffHook.1.contact=201/ /call This dials extension 201 when the handset is lifted. I think it's pretty self-explanatory, but if you have questions, feel free to ask... Best wishes, Jonathan Barratt Openface Internet Inc. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Thursday, June 21, 2007 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] hotline with Polycom Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0:number). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom? The idea is that you pick up the handset and it automatically dials a number. It will be used in a foyer or front door. Many thanks David. ___ --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] STDERR in AGI
Hi all,
I just started programming using AGI and I have a simple question about
STDERR.
If I understood it right, all the messages sent to STDERR should be
shown in the Asterisk console, but using the following python code I
just can't see anything.
#!/usr/bin/python
#
# File: /var/lig/asterisk/agi-bin/agi-test.py
#
# Description: An AGI Script
#
import sys
env = {}
tests = 0
while True:
line = sys.stdin.readline().strip()
if line == '':
break
key,data = line.split(':')
if key[:4] != 'agi_':
sys.stderr.write(Did not work!\n)
sys.stderr.flush()
continue
key = key.strip()
data = data.strip()
if key != '':
env[key] = data
sys.stderr.write(AGI Environment Dump:\n)
for key in env.keys():
sys.stderr.write( -- %s = %s\n % (key,env[key]))
sys.stderr.flush()
##
This code comes from the book Asterisk: The future of the Internet and
it is being activated by an extension like that:
exten = 123,1,Answer()
exten = 123,2,AGI(agi-test.py)
Any help would be appreciated.
Ronaldo.
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Re: [asterisk-users] VPN on Asterisk
i believe it also supports udp. however i don't use this. i was only stating that it is a solution that could work on some limited enviroiments (eg: only tcp/80 open orso). regards, --nvieira On Jun 18, 2007, at 10:28 PM, Eric ManxPower Wieling wrote: You do NOT want to send realtime audio over a TCP connection. Nuno Vieira - nfsi telecom wrote: try vtund. http://vtun.sourceforge.net/ its a userland tcp implementation... not the safest thing around, but should be secure enough for what you are looking for, and pretty simple to implement. cheers, --nvieira On Jun 18, 2007, at 7:37 PM, Remco Barendse wrote: Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. Slightly offtopic, but I would choose a VPN solution that can do webvpn (connect to port 80), i just came back from holiday and several hotels had VOIP *and* VPN ports for PPTP blocked in their internet, to prevent people from calling over their internet connection, clogging up their (pretty poor) connection. With webvpn you can connect to port 80 and circumvent such trouble. I tried finding an easy HOWTO for OpenVPN, on a CentOS box, this is not easy at all. ___ --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] Zaptel wct2xxp driver causes LEDs to go black and RED alarm
The problem was with ACPI screwing up interrupt routing. Added pci=routeirq to /boot/grub/grub.conf to turn off acpi for interrupt routing. Now I've got two green LEDs. Thanks to Jolan Luff for figuring this out! Jason :) On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wrote: Hi there, I've got two Asterisk hosted PBX servers with Digium TE210P cards attached by a E1 cable to Port 1 on each. On startup, both cards flash red, alternating between ports 1 and 2. When server #1 loads the Zaptel module and drivers, Port 1 status LED goes green. When server #2 loads the same module and drivers, Port 1 status LED goes completely blank. Unloading the wct2xxp module causes the flashing red LEDs to come back. I've tried swapping cable ends and cards between the two machines, but the problem LED always stays with server #2. So, I think there is something misconfigured with server #2, but the configuration file on both servers is identical. zaptel.conf: loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Any clues what could cause a GREEN alarm on one end with a RED alarm on the other and no LED light as soon as the wct2xxp driver is loaded? Thanks for the help, Jason Carter DLS Internet Services ___ --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 Faxing
Kyle, you are missing CPC on the line, asterisk is not detecting the hangup because your phone company is not giving it to you. Try busycount in zapata.conf On 6/21/07, Kyle Vorster [EMAIL PROTECTED] wrote: Any one know more about this, Please assist if possible. Kyle Vorster wrote: Any one able to assist, Please Paradise Dove wrote: so how to avoid CPC?? On 6/14/07, C F [EMAIL PROTECTED] wrote: Its called CPC On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote: Hello, Sorry if this is a real dumb question but when sending a fax and the end user does not enable fax on their side and then just hangs up does not force asterisk to end the call. So it keeps the trunk open until its killed by a Flash Operator. Please assist if any one understands me. Kind Regards, Kyle Virster ___ --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 ___ --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] different codec for different extensions
Hi Mojo, I dont have control our calling party. and also called extension is only configured in extensions.conf not sip.conf etc. So I must select the codec within my dialplan (extensions.com) I found one solution by using SIP_CODEC variable like [fax] exten = 605,1,ringing() exten = 605,n,set(SIP_CODEC=ulaw) exten = 605,n,RxFAX(/tmp/nasir.tiff|ecm) exten = 605,n,hangup() but Thanks for your answer Thanks Nasir Iqbal [userX] ... context=internal disallow=all allow=gsm allow=ulaw ... [fax] ... disallow=all allow=ulaw ... Then any IVRs that userX accesses should be in gsm because it's the preferred codec? Assuming that the gsm sound files ARE installed? You might experiment with this. But when userX is bridged to the fax channel, ulaw is the only one the fax channel allows, so it's chosen on both ends. Shouldn't this work? Mojo Nasir Iqbal wrote: Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --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 config files and #include
Yes I was aware of the MAX_INCLUDE_LEVEL define. Just wasnt sure about increasing it cos I thgt it might have been kept that low for a reason. I have my setup working perfectly fine right now ( I just reduced the number of files being included and there is no nesting either). Though I will try out your suggestions in the future when I need to make changes. Will let you know of my findings then. Thanks for your help. Regards, Deepak Tzafrir Cohen wrote: On Thu, Jun 21, 2007 at 04:07:03PM +0530, Deepak Bhat wrote: Im sure its not a circular include. Like you said its mostly realted to the number of nested includes but the exact meaning is not clear to me. I repeat: To trace this, enable debugging and debug logging. There is a debug comment for each included file. enable 'debug' for some log file in logger.conf , and then run: logger reload reload ___ --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] Once Touch Recording
Hi All, Once touch recording only seems to work between extensions. When calling an external party when pressing *1 does nothing. The person you have called can hear 2 DTMF tones. Is there a trick to getting once touch recording working over a zap channel? I am using a TE110P, but calls over SIP to a VSP also fails when trying to use one touch recording. ___ --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
