Re: [Asterisk-Users] ATA 186 MGCP Firmware
Hi Ken, Can't seem to find it anywhere, and my cisco login works, but says there's no longer any downloads available for the ATA186.. anyone know where I could find the MGCP version of the firmware via download? Log in. From the main page, click the dropdown list for Downloads and select Voice Software. That takes you to http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml Under Voice Applications Software, click on ATA 186/188 Analog Telephone Adaptor That took me to http://www.cisco.com/cgi-bin/tablebuild.pl/ata186 The latest seems to be ata_03_01_01_mgcp_040629_1.zip ATA Version 3.1.1 software for MGCP, 02-JUL-2004 When I clicked that link, the license agreement came up. I did not proceed, but it seems likely to work. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP vs cRTP vs IAX
Jean-Michel Hiver wrote: Hi List, I have seen this: http://www.convergence.com.pk/iax2/trunked.html According to this table, using trunking, you can have 16 channels with 171.7 kbps bandwith using g.729 + IAX2 trunking? Sounds too good to be true... Any comments on this? If I'm reading the little blue/green chart at the top correctly, the calculations on this page use some sort of special IP that only has 12 octets of header, instead of the 20 octets the IP standard specifies. So the sum of 20 octets of IP header plus 8 octets of UDP header would be 28 octets/224 bits, instead of the 160 bits shown there. They also show an RTP header size of 8 octets; I think it should be 12. Those would have an effect, I think, on all the derived calculations. I also see a descrepancy in the bitrate they use for iLBC. Depending on the sample size, its bitrate, according to the blurb on the front page of its website, is either 13.3kbs (30ms/sample) or 15.2 (20ms/sample). The convergence site states 9kbs. So either I've got some wrong info, or they do. . . . B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@home questions
Hi I am currently running [EMAIL PROTECTED] version 0.9 and have a few questions, which i hope someone on this list might be able to answer. 1) I am trying to setup incomming fax support, but however i never manage to receive the faxes, getting a signal 15. As per handbook, there isn't too much underlaying documentation, which one could reference to debug the problems. Here is a log: pr 24 03:01:01 NOTICE[1053]: Got event 2 (Ring/Answered)... Apr 24 03:01:02 DEBUG[1053]: Expression is '0' Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIf(40mZap/3-1, 0?from-pstn-reghours|s|1:) in new stack Apr 24 03:01:02 DEBUG[1053]: Not taking any branch Apr 24 03:01:02 DEBUG[1053]: Expression is '0' Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIf(40mZap/3-1, 0?from-pstn-afthours|s|1:) in new stack Apr 24 03:01:02 DEBUG[1053]: Not taking any branch Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIfTime(;35;40mZap/3-1, *|*|*|*?from-pstn-reghours|s|1:) in new stack Apr 24 03:01:02 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,1) Apr 24 03:01:02 DEBUG[1053]: Expression is '0' Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIf(40mZap/3-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack Apr 24 03:01:02 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,2) Apr 24 03:01:02 VERBOSE[1053]: -- Executing Answer(40mZap/3-1, ) in new stack Apr 24 03:01:02 DEBUG[1053]: Took Zap/3-1 off hook Apr 24 03:01:02 DEBUG[1053]: Enabled echo cancellation on channel 3 Apr 24 03:01:02 DEBUG[1053]: Engaged echo training on channel 3 Apr 24 03:01:02 VERBOSE[1053]: -- Executing Wait(mZap/3-1, 1) in new stack Apr 24 03:01:03 VERBOSE[1053]: -- Executing SetVar(40mZap/3-1, intype=GRP-1) in new stack Apr 24 03:01:03 VERBOSE[1053]: -- Executing Cut(Zap/3-1, intype=intype|-|1) in new stack Apr 24 03:01:03 DEBUG[1053]: Expression is '0' Apr 24 03:01:03 VERBOSE[1053]: -- Executing GotoIf(40mZap/3-1, 0?7:9) in new stack Apr 24 03:01:03 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,9) Apr 24 03:01:03 DEBUG[1053]: Expression is '1' Apr 24 03:01:03 VERBOSE[1053]: -- Executing GotoIf(40mZap/3-1, 1?10:12) in new stack Apr 24 03:01:03 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,10) Apr 24 03:01:03 VERBOSE[1053]: -- Executing Wait(mZap/3-1, 3) in new stack Apr 24 03:01:04 DEBUG[1053]: DTMF digit: f on Zap/3-1 Apr 24 03:01:04 VERBOSE[1053]: -- Redirecting Zap/3-1 to fax extension Apr 24 03:01:04 VERBOSE[1053]: == Spawn extension (from-pstn-reghours, fax, 0) exited non-zero on 'Zap/3-1' Apr 24 03:01:04 VERBOSE[1053]: -- Executing Goto(mZap/3-1, ext-fax|in_fax|1) in new stack Apr 24 03:01:04 VERBOSE[1053]: -- Goto (ext-fax,in_fax,1) Apr 24 03:01:04 DEBUG[1053]: Expression is '1' Apr 24 03:01:04 VERBOSE[1053]: -- Executing GotoIf(40mZap/3-1, 1?2:analog_fax|1) in new stack Apr 24 03:01:04 VERBOSE[1053]: -- Goto (ext-fax,in_fax,2) Apr 24 03:01:04 VERBOSE[1053]: -- Executing Macro(0mZap/3-1, faxreceive) in new stack Apr 24 03:01:04 VERBOSE[1053]: -- Executing SetVar(40mZap/3-1, FAXFILE=/var/spool/asterisk/fax/1114333259.710.tif) in new stack Apr 24 03:01:04 VERBOSE[1053]: -- Executing SetVar(40mZap/3-1, [EMAIL PROTECTED]) in new stack Apr 24 03:01:04 VERBOSE[1053]: -- Executing RxFAX(0mZap/3-1, /var/spool/asterisk/fax/1114333259.710.tif0m) in new stack Apr 24 03:02:15 DEBUG[1053]: Exception on 15, channel 3 Apr 24 03:02:15 DEBUG[1053]: Got event On hook(1) on channel 3 (index 0) Apr 24 03:02:15 DEBUG[1053]: disabled echo cancellation on channel 3 Apr 24 03:02:15 DEBUG[1053]: Got hangup Apr 24 03:02:15 DEBUG[1053]: Extension s, priority 3 returned normally even though call was hung up Apr 24 03:02:15 DEBUG[1053]: Extension in_fax, priority 2 returned normally even though call was hung up Apr 24 03:02:15 VERBOSE[1053]: -- Executing Hangup(40mZap/3-1, ) in new stack Apr 24 03:02:15 VERBOSE[1053]: == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/3-1' Thus as one sees it seems i get a exception. I did do the asterisk pdf install. Is there any way to get better debugging info? Also any way to configure asterisk to keep the tiff files around and not delete them when they are sent? 2) The next question i have is how do i manipulate a physical line via asterisk on a sip client. Basically i am subscribed to some services via the telco like call waiting / call forwarding, where as using the *70 on a sip will just do a call forwarding on asterisk and not send that signal out to the telco. Also trying to manipulate the call waiting signal on the line, as when i am on the sip phone i can hear a call come in, but can't seem to instruct asterisk to send the switch voice channel signal to the telco. Any ideas on how to do this? Not in the documentation from what i can read. Please let me know Thanks Sascha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] X100P Clone any hints for recognizing RINGing ?
On Mon, 25 Apr 2005, Andrew Kohlsmith wrote: It has absolutely nothing to do with what economically suits them best -- it has everything to do with the fact that when you buy a clone X100P you DO NOT KNOW what you're getting. The chipset may be the same but as you can clearly see from searching this very list, the hybrid circuitry (a crucial crucial part of the design) can be VERY different, and even if the hybrid's fine, there are subtle variations in the chipset that can bite you in the ass. Hi everyone, I made the same mistake and bought a cheap clone. I do understand that these cheap X100P cards do not work well enogh for production use. But as I already bought this card, can anyone give me some advice regarding the following: Dialing out from Asterisk works like a charm, but recognizing a RING is much more different. It works in 1 of 10 cases that Asterisk recognizes the ringing on the phone :-) Where could I change which parameters to make Asterisk a little more sensible ? Please help and do not tell me to buy another card - I do understand what I bought, but now I already have it and want to use it. Thanks a lot! Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX aproprietary protocol
Joseph wrote: Can anybody explain me why IAX is called proprietary protocol? In some places IAX is refereed as open protocol. How can proprietary protocol be open protocol? Since the source code is available to anyone and GPL'ed it is an open protocol. However it's not a standard and there is no clear specification other than the source code (which kind of sucks because it means that ATA makers probably won't support it). So it's a non-standard, open protocol. Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Clone any hints for recognizing RINGing ?
Hi Chris, I've had A LOT of experience with the cheap X100Ps in the last few weeks. I myself bought two of them off ebay. $6.95 special! I also had problems with them and within the past 12 hours have replaced them with a TDM22B that so far(1 phone call) has worked great. I would suggest turning verbosity on and watching the asterisk console when the call comes in. If it is not showing that there is a ring on your Zap device, then it's not a problem with your dial plan. I would say that it's likely that the modem would be defective if sometimes it detected a ring and other times not. If asterisk is always showing that there is a ring, then I would take another good look at the context in your extensions.conf that matches the one set for your zap device in Zapata.conf. While trying to sort out my X100P clone problem's on here, a user rudely told me to buy a real card. Ended up being good advice, just not tactful. I could have saved WEEKS of headaches had I not tried to save some $$. I guess I learned the hard way that you get what you pay for. I'm not telling you to buy another card, I'm just sharing my experience with X100P. Hope this helps, Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Rothe Sent: Thursday, April 28, 2005 01:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] X100P Clone any hints for recognizing RINGing ? On Mon, 25 Apr 2005, Andrew Kohlsmith wrote: It has absolutely nothing to do with what economically suits them best -- it has everything to do with the fact that when you buy a clone X100P you DO NOT KNOW what you're getting. The chipset may be the same but as you can clearly see from searching this very list, the hybrid circuitry (a crucial crucial part of the design) can be VERY different, and even if the hybrid's fine, there are subtle variations in the chipset that can bite you in the ass. Hi everyone, I made the same mistake and bought a cheap clone. I do understand that these cheap X100P cards do not work well enogh for production use. But as I already bought this card, can anyone give me some advice regarding the following: Dialing out from Asterisk works like a charm, but recognizing a RING is much more different. It works in 1 of 10 cases that Asterisk recognizes the ringing on the phone :-) Where could I change which parameters to make Asterisk a little more sensible ? Please help and do not tell me to buy another card - I do understand what I bought, but now I already have it and want to use it. Thanks a lot! Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux SoftPhone with Sound Daemon Support
Does anyone know of a Linux SoftPhone that will play nicely with ESD? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridialplan/TON question
Hi Peter! FYI: Yesterday i put Asterisk between a Hicom 350E and a Telekom Austria (TA) PRI. Both use TON=unknown for called number, but Hicom always uses TON=international for calling number whereas TA uses a dynamic TON for calling number. Thus, for incoming calls (PSTN-PBX) the presented caller number will be incorrect = CALLINGTON is needed to fix this. Peter, please send me the patch. regards, Klaus Peter Svensson wrote: On Tue, 26 Apr 2005, Marc Storck wrote: I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? CALLINGTON was not populated in -stable. Tha patch was only added to -head. It is not that hard to add, I can send you our old patch if you want it. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, Asterisk and NAT
As mentioned yesterday, i made an attempt to write documentation to get NAT + SIP to work on http://www.asteriskguru.com/natut.php If you send me the info for those phones, firewalls i will include them. (I was planning on adding some Linux/BSD firewall rules but i dont have a pix,). /Z Irakli Natsvlishvili wrote: Hi there, There are plenty of good documents on Asterisk, SIP and NAT on the voip-info.org wiki. Please look them up. There are also information within the configs/sip.conf.sample file within Asterisk. Folks, let's face it - documentation on Asterisk sucks big time. This is the reason why the same questions are asked here and over the Net every week. If Asterisk is on a public IP, again: it's up to the phones. It's still not an Asterisk problem. Yes, but you need to pick the right phone, the right NAT/FW and have a lot of patience :-) OK, let's document is. Is there any information with different phones/FW combinations already available? I can add some info working with Cisco's IP phones, Pix firewall, cheap linksys/dlink gateways. Good NAT traversal support. and we do not give them any NAT traversal support. Why? Is this for some political reasons? Irakli ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work withasterisk!
Hello Tomek, I also got a diva pci 2.02 card, but although the kernel sees the incoming calls, asterisk refuses to answer. Did you have this issue at all? The kernel seems to be denying the call... Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Wednesday, March 23, 2005 11:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work withasterisk! I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not server divas) with asterisk. First thing to do is to load the module. If you have two of these cards, you should do it like that: modprobe -v hisax protocol=2,2 type=11,11 And now you can have up to 4 incoming calls with two cards (try calling yourself and see if anything gets into your syslog - you should have ignored calls even if asterisk isn't running). Then configure your asterisk to use i4l (don't use chan_capi) - do it in modem.conf: (...) driver=i4l (...) msn=your_msn_number and that's it (you still need to configure your ISDN devices to allow incoming calls, for example, using conf-isdn-account - don't forget to set SECURE=off etc. ISDN settings). Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX ATA's
I would also be interested in a multi-port ata that supports iax. The only single port ata I know of (besides the iaxy) that supports iax is the PA168 from china. cheers Clive On 27 Apr 2005 at 11:15, Rod Bacon wrote: What sort of price are they asking for a 4-port gateway? - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 27, 2005 8:53 AM Subject: Re: [Asterisk-Users] IAX ATA's There is Taiwan company Soundwin that seems to me are willing to support asterisk protocol in their equipment. I was looking for 1FXO x 3-4FXS in one unit. http://www.soundwin.com/ I just exchanged few email with Sam at [EMAIL PROTECTED] and was able to convince them to add support for IAX2; they seems to me listen to the end user so I suggest some of you drop him an email and express your interest in their product if they will support asterisk protocol. --quote We have plan to implement IAX or IAX2 in our product line including 2 -8 port VoIP Gateway in Q3. Thanks your information and we would pay more attention in Asterisk community. -end quote- -- #Joseph On Tue, 2005-04-26 at 16:46 -0400, Garrett Smith wrote: Does anyone know of a quality alternative to the Digium IAXy? I have a customer experiencing numerous issues such as over heating with the older IAXyÿs and the new IAXy is not yet available. Can anyone recommend an alternative? Thanks, Garrett Smith [EMAIL PROTECTED] B2 Technologies/ VoIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and more. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 186 MGCP Firmware
I've been there.. the page comes up with There are currently no files for this type. :( k Stewart Nelson wrote: Hi Ken, Can't seem to find it anywhere, and my cisco login works, but says there's no longer any downloads available for the ATA186.. anyone know where I could find the MGCP version of the firmware via download? Log in. From the main page, click the dropdown list for Downloads and select Voice Software. That takes you to http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml Under Voice Applications Software, click on ATA 186/188 Analog Telephone Adaptor That took me to http://www.cisco.com/cgi-bin/tablebuild.pl/ata186 The latest seems to be ata_03_01_01_mgcp_040629_1.zip ATA Version 3.1.1 software for MGCP, 02-JUL-2004 When I clicked that link, the license agreement came up. I did not proceed, but it seems likely to work. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Panasonic KX-TD1232 Signaling
On Wed, 27 Apr 2005, Dan Morin wrote: To expand upon my original question, does anyone know of any devices that would make connectivity between the Panasonic system and Asterisk possible? What are opinions of using FXS ports in Asterisk going into to CO ports on the PBX? Or if I'm putting money into the problem, how would I go about setting up a PRI connection between the two? A PRI connection would give me 20 odd lines correct? What kind of CO lines does your current setup use? Standard pots lines, cahnneliazed (non-isdn) T1 with rbs, isdn bri, tie-lines or something else? A T1 PRI card will give you 23 lines between the panasonic KX-TD1232 and Asterisk, an E1 PRI card will give you 30 lines. BRI or PRI is really the best way of interconnecting Asterisk and the pbx. A new PRI card should cost about $1000-$2000. Make sure you get a PRI T1 card and not a non-isdn T1 card. If the KX-TD1232 uses BRI CO lines they can be used instead. It may be hard to obtain BRI cards in the USA. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: UK (english) sound files (Paul R)
So now that they are done how about you post the files for us? Share the wealth. Mark Will be happy to do so once macro refined a little, but it is rather long (about 600 lines) and I thought long posts were bad manners. Otherwise this will be odne by the end of the weekend/ Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work withasterisk!
Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, I also got a diva pci 2.02 card, but although the kernel sees the incoming calls, asterisk refuses to answer. Did you have this issue at all? The kernel seems to be denying the call... if you see the calling in the systlog, that's 98% of success :) you have to set up in modem.conf something like: driver=i4l ; your msn - without it (or if it's wrong) it won't work msn=4235 device = /dev/ttyI0 device = /dev/ttyI1 restart asterisk, and it should pick up the phone now (or, you don't have it configured in asterisk, but the default configuration should pick up the phone and play a demo). check asterisk logs if it sees an incoming call. Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel FXO crashing.
Jason Leach wrote: About every 24-48h the Zaptel FXO port crashes. If I pick up my phone and try to make a call on the FXS port I get a hissing and squealing sound. Seems to be right where Asterisk makes the bridge. Also Asterisk does not answer an inbound call on the FXO port; does not even display as ringing. To get the system working again. I must stop asterisk, restart zaptel and then restart asterisk. Next time an FXO stops responding, stop asterisk and do a register dump of the offending module. You may need to cd into the zaptel src directory - I'm not sure that fxstest is installed. ./fxstest /dev/zap/1 regdump will show you the contents of all the registers on Zap 1. If the majority of them show the value ff, contact Digium support. I had modules marked Rev C that did this replaced with X100B RevB ones and have not had any trouble since. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call recording problem
It seems that there areoccasional problems with files generated by soxmix utility. The Asterisk console would show the following message: soxmix: Overriding output size to bytes for compressed data soxmix: help! internal inconsistency - data_written 12156 gsmbytecount 12155. When trying to play this file using Windows media player, an error message will pop up while playing the end of the file. I'm using wav49 format, but I found the same problem would occur for wav format. Does anybody encounter the same problem? Under which circumstances will this problem occurs? I'm running it on Fedora 2. Please advise. Thanks. Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) workwithasterisk!
Hello Tomek, Previously I did get asterisk to see the call, but not currently. This is in the usa, so my msn is a 7 digit number. The kernel is saying the following: Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 - 2781980 Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511 - 0 2781980 ignored Apr 28 03:46:17 localhost kernel: isdn_tty: call from 8005966511 - 2781980 ignored Apr 28 03:46:20 localhost kernel: isdn_net: call from 8005966511,1,0 - 2781980 Apr 28 03:46:20 localhost kernel: isdn_net: call from 8005966511 - 0 2781980 ignored Apr 28 03:46:20 localhost kernel: isdn_tty: call from 8005966511 - 2781980 ignored Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Thursday, April 28, 2005 3:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) workwithasterisk! Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, I also got a diva pci 2.02 card, but although the kernel sees the incoming calls, asterisk refuses to answer. Did you have this issue at all? The kernel seems to be denying the call... if you see the calling in the systlog, that's 98% of success :) you have to set up in modem.conf something like: driver=i4l ; your msn - without it (or if it's wrong) it won't work msn=4235 device = /dev/ttyI0 device = /dev/ttyI1 restart asterisk, and it should pick up the phone now (or, you don't have it configured in asterisk, but the default configuration should pick up the phone and play a demo). check asterisk logs if it sees an incoming call. Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QuadBRI card on Suse 9.2 Unable to load qozap.ko
Massimo wrote: Hi, I successfully installed zaptel,libpri,asterisk and qozap in a Suse 9.2. I removed the old modules loaded as default by Suse. Now I'm triying to load qozap.ko but I receive this error: Did you do the install with bristuff-0.2.0-RC8a ? Works nice on debian, I guess it will be the same on Suse. Cheers, Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) workwithasterisk!
Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, Previously I did get asterisk to see the call, but not currently. This is in the usa, so my msn is a 7 digit number. The kernel is saying the following: Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 - 2781980 so this is your MSN: 2781980 try loading the default asterisk config files, and you should be able to use your card. or try [EMAIL PROTECTED] - asteriskathome.sf.net - it's asterisk made easy (well, sort of) - if you decide to use it, let me know, because Eicon cards won't work with it right after installation (you have to yum install kernel-unsupported etc., then load hisax module etc.) Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver) workwithasterisk!
Hello Tomek, When I call my second msdn, I get the following: == Starting Modem[i4l]/ttyI1 at incoming-isdn,2781984,1 failed so falling back to exten 's' -- Executing Answer(Modem[i4l]/ttyI1, ) in new stack somersvoip*CLI Apr 28 04:04:33 localhost kernel: isdn_net: call from 8005966511,1,0 - 2781984 Apr 28 04:04:33 localhost kernel: isdn_net: call from 8005966511 - 0 2781984 ignored Apr 28 04:04:33 localhost kernel: isdn_tty: call from 8005966511, - RING on ttyI1 Apr 28 04:04:33 localhost kernel: isdn: HiSax,ch0 cause: E0260 Apr 28 04:04:43 WARNING[4476]: chan_modem_i4l.c:555 i4l_answer: Unable to answer: NO CARRIER == Spawn extension (incoming-isdn, s, 1) exited non-zero on 'Modem[i4l]/ttyI1' -- Hungup 'Modem[i4l]/ttyI1' I will try the defaults at some point this week, it's at the other office. Hopefully I'll make out ok... Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Thursday, April 28, 2005 3:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver) workwithasterisk! Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, Previously I did get asterisk to see the call, but not currently. This is in the usa, so my msn is a 7 digit number. The kernel is saying the following: Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 - 2781980 so this is your MSN: 2781980 try loading the default asterisk config files, and you should be able to use your card. or try [EMAIL PROTECTED] - asteriskathome.sf.net - it's asterisk made easy (well, sort of) - if you decide to use it, let me know, because Eicon cards won't work with it right after installation (you have to yum install kernel-unsupported etc., then load hisax module etc.) Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver) workwithasterisk!
Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, When I call my second msdn, I get the following: == Starting Modem[i4l]/ttyI1 at incoming-isdn,2781984,1 failed so falling back to exten 's' -- Executing Answer(Modem[i4l]/ttyI1, ) in new stack somersvoip*CLI Apr 28 04:04:33 localhost kernel: isdn_net: call from 8005966511,1,0 - 2781984 Apr 28 04:04:33 localhost kernel: isdn_net: call from 8005966511 - 0 2781984 ignored Apr 28 04:04:33 localhost kernel: isdn_tty: call from 8005966511, - RING on ttyI1 Apr 28 04:04:33 localhost kernel: isdn: HiSax,ch0 cause: E0260 Apr 28 04:04:43 WARNING[4476]: chan_modem_i4l.c:555 i4l_answer: Unable to answer: NO CARRIER == Spawn extension (incoming-isdn, s, 1) exited non-zero on 'Modem[i4l]/ttyI1' -- Hungup 'Modem[i4l]/ttyI1' I will try the defaults at some point this week, it's at the other office. Hopefully I'll make out ok... yeah try the defaults first. I would look what Unable to answer: NO CARRIER means if the defaults won't work. Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls and CAPI
Hello all, I just instaled an AVM C4 card on my asterisk to connect it to the PSTN and send or revibe calls using it. I can make calls perfectly, two calls over the same port at same time. The problem appears when a call arribes. CAPI seems to answer it and pass it to asterisk and the call hangs up. Any clue will be welcomed. thaks for your time. I get this logs errors. -- CONNECT_IND ID=001 #0x0bce LEN=0035 Controller/PLCI/NCCI = 0x102 CIPValue= 0x1 CalledPartyNumber= 89402 CallingPartyNumber = 09 80201 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default Apr 28 05:05:45 NOTICE[727]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=001 #0x0bce LEN=0035 Controller/PLCI/NCCI = 0x102 CIPValue= 0x1 CalledPartyNumber= 89402 CallingPartyNumber = 09 80201 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default == CONNECT_IND (PLCI=0x102,DID=402,CID=201,CIP=0x1,CONTROLLER=0x2) == Starting CAPI[contr2/402]/18 at llamadas_entrantes,402,1 failed so falling back to exten 's' -- Executing GotoIf(CAPI[contr2/402]/18, 0?horario_oficina_okm|s|1) in new stack -- Executing GotoIf(CAPI[contr2/402]/18, 0?horario_oficina_alaire|s|1) in new stack -- Executing GotoIf(CAPI[contr2/402]/18, 0?llamada_aitor_gil|s|1) in new stack -- Executing GotoIf(CAPI[contr2/402]/18, 0?contexto_fax_recepcion1|s|1) in new stack -- Executing Goto(CAPI[contr2/402]/18, contexto_operadora_okm|s|1) in new stack -- Goto (contexto_operadora_okm,s,1) -- Executing Answer(CAPI[contr2/402]/18, ) in new stack -- Executing SetLanguage(CAPI[contr2/402]/18, es) in new stack -- Executing Dial(CAPI[contr2/402]/18, SIP/100|70|Ttr) in new stack -- Called 100 -- started pbx on channel (callgroup=2)! -- INFO_IND ID=001 #0x0bcf LEN=0019 Controller/PLCI/NCCI = 0x102 InfoNumber = 0x70 InfoElement = 89402 -- INFO_IND ID=001 #0x0bd0 LEN=0016 Controller/PLCI/NCCI = 0x102 InfoNumber = 0x18 InfoElement = 89 -- ALERT_CONF ID=001 #0x0bce LEN=0014 Controller/PLCI/NCCI = 0x102 Info = 0x0 -- INFO_IND ID=001 #0x0bd1 LEN=0017 Controller/PLCI/NCCI = 0x102 InfoNumber = 0x8 InfoElement = 82 90 -- DISCONNECT_IND ID=001 #0x0bd2 LEN=0014 Controller/PLCI/NCCI = 0x102 Reason = 0x3490 == DISCONNECT_IND PLCI=0x102 REASON=0x3490 == Spawn extension (contexto_operadora_okm, s, 3) exited non-zero on 'CAPI[contr2/402]/18' -- CONNECT_IND ID=001 #0x0bd3 LEN=0035 Controller/PLCI/NCCI = 0x102 CIPValue= 0x1 CalledPartyNumber= 89409 CallingPartyNumber = 09 80201 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default Apr 28 05:05:47 NOTICE[727]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=001 #0x0bd3 LEN=0035 Controller/PLCI/NCCI = 0x102 CIPValue= 0x1 CalledPartyNumber= 89409 CallingPartyNumber = 09 80201 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default == CONNECT_IND (PLCI=0x102,DID=409,CID=201,CIP=0x1,CONTROLLER=0x2) == Starting CAPI[contr2/409]/19 at llamadas_entrantes,409,1 failed so falling back to exten 's' -- Executing GotoIf(CAPI[contr2/409]/19, 0?horario_oficina_okm|s|1) in new stack -- Executing GotoIf(CAPI[contr2/409]/19, 0?horario_oficina_alaire|s|1) in new stack -- Executing GotoIf(CAPI[contr2/409]/19, 0?llamada_aitor_gil|s|1) in new stack -- Executing GotoIf(CAPI[contr2/409]/19, 0?contexto_fax_recepcion1|s|1) in new stack -- Executing Goto(CAPI[contr2/409]/19, contexto_operadora_okm|s|1) in new stack -- Goto (contexto_operadora_okm,s,1) -- Executing Answer(CAPI[contr2/409]/19, ) in new stack -- Executing SetLanguage(CAPI[contr2/409]/19, es) in new stack -- Executing Dial(CAPI[contr2/409]/19, SIP/100|70|Ttr) in new stack -- Called 100 -- started pbx on channel (callgroup=2)! -- INFO_IND ID=001 #0x0bd4 LEN=0019 Controller/PLCI/NCCI = 0x102 InfoNumber = 0x70 InfoElement = 89409 -- INFO_IND ID=001 #0x0bd5 LEN=0016 Controller/PLCI/NCCI = 0x102 InfoNumber = 0x18 InfoElement = 89 -- ALERT_CONF ID=001 #0x0bd3 LEN=0014 Controller/PLCI/NCCI = 0x102 Info = 0x0 -- INFO_IND ID=001 #0x0bd6 LEN=0017 Controller/PLCI/NCCI = 0x102 InfoNumber = 0x8 InfoElement = 82 90 -- DISCONNECT_IND
Re: [Asterisk-Users] * and Sipgate (UK)
Luki wrote: Robert, It looks like you're dialing 447733322998, 44 for UK, then the area code, etc. I have sipgate.de setup to dial local numbers (any German number) as 0+AREA CODE+NUMBER. Always dial the area code, even if you sipgate number is in the same city. For international numbers you need to dial 00+COUNTRY CODE+AREA CODE+NUMBER. I think similar rules apply for sipate.co.uk, so try dialing the above as: 07733322998 or 00447733322998. Doh, I had tried several combinations of dailing, however I didn't try just 077xxx that worked fine. I thought it was the way I was dailing as other ways I'd tried had failed. Thats got it working. Thanks for the wake up :) Besides that, maybe a stupid question, but do you have money in your sipgate account? Yeah :) -- Robert P. McKenzie | GammaRay Technical Services Ltd [EMAIL PROTECTED] | [EMAIL PROTECTED] http://www.uk-experience.com | http://www.gammaray-tech.com Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring B chans and G.729 High Water Marks
In capacity planning for production Asterisk servers it is essential to have an accurate statistical picture of the utilisation of finite resources such as disk space, CPU utilisation, B channels on PRIs, and G.729 codec licences. The first two have well defined measurement tools. The last two (B channels and G.729) can be measured in real time using show channels and show g729 but there appears no way to obtain from Asterisk the maximum number of B channels that have been in use or the maximum number of g729 codecs that have been in use. Also there seem not to be any tools for collecting this information over time for statistical analysis. What do people do to monitor increases in channels/g729 licences so to plan ordering more of each? Thanks for your suggestions. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 Zone
Sebastian Atala wrote: Hi, Someone knows how can I register my Asterisk to a gatekeeper using zone parameters? I'm using asterisk 1.0.7 and oh323 0.6.5. I'm trying to register to a gatekeeper in another network and I can't reach this with a broadcast. Zone is the name who Cisco call the GK identification. In oh323.conf set: gatekeeper=GKID:zone_name e.g. if lala is the zone name: gatekeeper=GKID:lala Thank in advance SA Michael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with X101P(Red Alarm)
I have bought some Wildcard X101P and Generic Clones for my Asterisk PBX. Now I can place and get calls through the lines/channels. Everything is okay but the problem is when I call outside through our PSTN line, after few minutes the connection breaks down. The same thing happens in case of incoming calls. I have checked my wiring and don't face that problem using direct connection. Whenever I call using that card, after few minutes I get a RED Alarm and if I reconnect the line, the Alarm is cleared. Therefore, I cannot continue my conversation through that line. Can anybody help me regarding this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
hello,i'm a naw asterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone answer the call and i've echo please may somebody help me?
there is my config file
zaptel.conf
fxsls=4#X100Pdefaultzone=frloadzone=fr
zapata.conf
[channels]language=fr relaxdtmf=yesimmediate=nocontext=pstnsignalling=fxs_ls;X100P;Cidsignalling=v23;Cidstart=polarity;usecallerid=yes;callerid="fone" 60
extensions.conf
[general]static=yeswriteprotect=no
[pstn]exten = 19100,1,dial(SIP/799SIP/788)
exten = 788,1,dial(SIP/788:5060)
exten = 799,1,dial(SIP/799:5060)
exten = _00N,1,dial(Zap/4/${EXTEN:1}); i want to call analog phone
exten = _6059,1,dial(SIP/799)exten = s,1,dial(SIP/799SIP/788);here i can recieve analog calls
regards.
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[Asterisk-Users] sip and analog
hello,i'm a naw asterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone answer the call and i've echo please may somebody help me?
there is my config file
zaptel.conf
fxsls=4#X100Pdefaultzone=frloadzone=fr
zapata.conf
[channels]language=fr relaxdtmf=yesimmediate=nocontext=pstnsignalling=fxs_ls;X100P;Cidsignalling=v23;Cidstart=polarity;usecallerid=yes;callerid="fone" 60
extensions.conf
[general]static=yeswriteprotect=no
[pstn]exten = 19100,1,dial(SIP/799SIP/788)
exten = 788,1,dial(SIP/788:5060)
exten = 799,1,dial(SIP/799:5060)
exten = _00N,1,dial(Zap/4/${EXTEN:1}); i want to call analog phone
exten = _6059,1,dial(SIP/799)exten = s,1,dial(SIP/799SIP/788);here i can recieve analog calls
regards.
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[Asterisk-Users] H323 FAX
hello i successfully installed asterisk on fedora core 3 and all what's in the check list plus the ACTOS gui and asterisk manager but i used actos to configure my cisco ip phones and dial/receive calls through sip. my problem is i need to configure H323/Fax in asterisk to catch H323/Fax from the gateway and route it as t38/fax to another pbx server i installed on windows. how can i configure, route and convert the faxes? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960s and skinny
Hi Paul, I had the same situation - I had a 7940 with only the callmanager firmware but would have much rathered SIP. You need to have a support contrace with Cisco to be able to download the firmware from their website. Thankfully the support contract only costs about $9 for the year - I was able to buy the contract from CDW ([EMAIL PROTECTED]) - the product code for the contract is CON-SNT-CP7940 (replace the 7940 at the end with 7960 for your phone). I'm in Ireland so it seems there is no problem purchasing internationally either. Derek Paul wrote: Do you still have that image for the 7960? I bought a 7940 on ebay and it doesn't have the SIP firmware. I can't find it anywhere but Cisco's website and they require that I have an account with them. Did you happen to save that binary file? Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton Sent: Tuesday, April 12, 2005 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960s and skinny Simon: I have had Skinny going on a 7960 (which I then reimaged to SIP). I currently run a 7910 on Skinny (using chan_sccp) and use the aforementioned 7960 simultaneously. Since you mentioned that you will have 50 phones, I assume you are using them in a business setting. I would *highly* recommend using SIP, as I have found that the skinny driver is not as reliable as it could be (not criticizing Jan or Julien at all, here). Reimaging the 50 of them should only take a while (depending on what version of CCM they have at the moment). I reimaged 12 phones once for a business and it took less than 30 minutes after I got it going (toying with the phones to get them to take the image, exactly how the config files were to be set up, etc...). I imagine you could easily get the whole thing done in less than a day (reimaging and config files), then figure out your dialplan. Then there is the whole issue of writing the config files...but you'd have to do those with Skinny, anyhow. I think with SIP you'll have much better reliability. -Andy FWD: 428725 On Apr 12, 2005 12:48 PM, Morris, Simon [EMAIL PROTECTED] wrote: Hello, Does anyone else have * running with Cisco 7960 phones and skinny? All the advise I am reading so far is telling me to load the SIP image on the phone but I'd like to know what I'm going to lose by persisting with skinny (Not reimaging 50 phones is one benefit amongst others of skinny) Thanks for any comparisons you can provide Rgds ~sm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with skinny
I have a couple of Cisco 7910's and I'd like to get them working with asterisk. I have two X101P wild cards installed and they are functioning well (Other two cards showing Red Alarm after few minutes conversation). I have configured foure cisco 7960 with SIP and they are working fine with *.I have been trying for a week to make the 7910's to work with * by skinny protocol but I am tired to tuning them. I already followed the instructions found in that list but still unsuccessful. Can anybody guide me to make them work? My problem is, When I plug the phone in, Phone shows: -- Configuring VLAN / -- Configuring IP / -- Opening my * server's IP and at that moment, * shows: -- Starting Skinny session from IP ADDRESS of MY PHONE Device SEP00044D076874 is attempting to register -- Device 'test' successfuly registered Requesting capabilities Received CapabilitiesRes RECEIVED UNKNOWN MESSAGE TYPE: 2b Buttontemplate requested Sending default template to [EMAIL PROTECTED] () Recieved SoftKey Template Request Received SoftKeySetReq Received LineStateReq Ouch ... error while writing audio data: : Broken pipe Segmentation fault And * is automatically stopped. Therefore I have to unpluge the phone and start the * again. Here is my * configuration files skinny.conf contains for cisco 7910 IP phone: [general] port = 2000 bindaddr = my * server's IP dateFormat = M-D-Y keepAlive = 120 [test] device=SEP00044D076874 version=P00405000600 context=default nat=0 callwaiting=1 transfer=1 threewaycalling=1 line = 800 extensions.conf contains for cisco 7910 IP phone: exten = 800,1,Dial(SKINNY/[EMAIL PROTECTED]|25|rt) Please help me to solve the problem. I appreciate both the magnitude of your appreciation and that of your support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960s and skinny
i have the cisco 7940 7960 phones and both i can use sip with asterisk? how can i help? On Thu, 2005-04-28 at 10:48 +0100, Derek Conniffe wrote: Hi Paul, I had the same situation - I had a 7940 with only the callmanager firmware but would have much rathered SIP. You need to have a support contrace with Cisco to be able to download the firmware from their website. Thankfully the support contract only costs about $9 for the year - I was able to buy the contract from CDW ([EMAIL PROTECTED]) - the product code for the contract is CON-SNT-CP7940 (replace the 7940 at the end with 7960 for your phone). I'm in Ireland so it seems there is no problem purchasing internationally either. Derek Paul wrote: Do you still have that image for the 7960? I bought a 7940 on ebay and it doesn't have the SIP firmware. I can't find it anywhere but Cisco's website and they require that I have an account with them. Did you happen to save that binary file? Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton Sent: Tuesday, April 12, 2005 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960s and skinny Simon: I have had Skinny going on a 7960 (which I then reimaged to SIP). I currently run a 7910 on Skinny (using chan_sccp) and use the aforementioned 7960 simultaneously. Since you mentioned that you will have 50 phones, I assume you are using them in a business setting. I would *highly* recommend using SIP, as I have found that the skinny driver is not as reliable as it could be (not criticizing Jan or Julien at all, here). Reimaging the 50 of them should only take a while (depending on what version of CCM they have at the moment). I reimaged 12 phones once for a business and it took less than 30 minutes after I got it going (toying with the phones to get them to take the image, exactly how the config files were to be set up, etc...). I imagine you could easily get the whole thing done in less than a day (reimaging and config files), then figure out your dialplan. Then there is the whole issue of writing the config files...but you'd have to do those with Skinny, anyhow. I think with SIP you'll have much better reliability. -Andy FWD: 428725 On Apr 12, 2005 12:48 PM, Morris, Simon [EMAIL PROTECTED] wrote: Hello, Does anyone else have * running with Cisco 7960 phones and skinny? All the advise I am reading so far is telling me to load the SIP image on the phone but I'd like to know what I'm going to lose by persisting with skinny (Not reimaging 50 phones is one benefit amongst others of skinny) Thanks for any comparisons you can provide Rgds ~sm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is There Media Accelerator For Better Asterisk Calls
--- Matt Riddell [EMAIL PROTECTED] wrote: In order to try and confirm this, see if small packets get loss. There was a packet loss in all the codecs i used, but it's hard to tell whether the packet percentage loss is gretaer or less at different codecs and how that help in solving the issue. The other option (if you have access to the egress point - router etc) My * box is behind the nat from a small isp and i don't have access to anything besides my box. I appreciate your feedback about IP compression. If all my voice calls' destinations are known, what can I do to reduce the ip header bandwidth. That might not help me here, but it will in other type of internet providers. The other option I like to try is to to install some accelerator to make more bandwidth available to my asterisk box. I appreciate any more suggestions. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RSS feed Asterisk-Users
hello Asterisk-Users I created just for fun a rss feed for Asterisk-Users and Asterisk-Biz list First for myself but if it is usefull for you can use it if you like. But some questions Is it allowed ? If you need add-on's please let me know. When many people will use it I need to generate a little money to pay traffic is it allowed to ad googleadd's ? I've added a search box and later on I will add the whole list history so it will be usefull to search. Just look at http://asterisk.voipexco.com Thanks, and have fun Sjaak ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel FXO crashing.
On April 27, 2005 01:25 am, Jason Leach wrote: hi, I have one of the latest versions of Asterisk CVS' (1.0.7-x) and the accompanying Zaptel drivers. The zaptel drivers are for my TDM400P w/ 1FXS and 1FSO. It all runs on CentOS 4.0 and a Dell Precision 410 w/ Dual PIII 700Mhz CPUs. About every 24-48h the Zaptel FXO port crashes. If I pick up my phone and try to make a call on the FXS port I get a hissing and squealing sound. Seems to be right where Asterisk makes the bridge. Also Asterisk does not answer an inbound call on the FXO port; does not even display as ringing. Sounds like you should call Digium for some technical support; you are entitled to support from them when you buy their products; that is what a lot of people don't seem to realize, and it certainly sounds like a card problem. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
Ok... so I can safely use a provider who uses G729 or G723 to provide me with VoIP termination without either A) having a connectivity issue, or B) having to pay trans-coding license costs? If so... what do I need to do to get asterisk to use G723? Just set it up? Am I going to have an issue if my terminator uses G723 as far as getting to/from phones using G726 or G711u? Or if someone calls in to check voicemail... will they not be able to hear anything because of this problem? What do I need to do to hook up to a G729 provider, and (the same question)... will a G711 phone work correctly (connected to asterisk). On 4/27/05, Rusty Shackleford [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, April 27, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Confused on G723 and G729 My question is.. if my voip terminator supports G723 and G729 only, do I still need a license? Or is that considered pass-through? If so, do I need to do anything special to get it to work? It is pass-through if both end points are using G.729. You need a G.729 license for every instance where a G.729 stream is encoded or decoded on your box. If you connect G.729 endpoints together, this isn't happening so no license is needed. Same goes for G.723. I'm also a litle confused about why G723 can do pass-through but can't do voicemail access? There is no G.723 license available for asterisk, ergo no way to transcode the voicemail and other promts into that format. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 04/25/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
For instance.. when I try to use G723.1 on my phone (and just call in from my PRI line) I get: Unable to find a path from g723 to ulaw. Unable to find a path from ulaw to g723. No path to translate from Zap/1-1(68) to Sip/201-80c7(1). Same things happens if I call in on my current provider's number which uses G711 for the codec. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP - capi problem (no sound)
Hi, After a kernel downgrade to 2.6.10 instead of 2.6.11.7 its working correctly. Sincerely, Cyrille De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cyrille Demaret Envoy: jeudi 28 avril 2005 0:00 : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet: [Asterisk-Users] SIP - capi problem (no sound) Hi, Ive a problem with the capi channel. Ive a SIP phone and a CAPI card configured in asterisk. CAPI - SIP: working SIP - SIP: working SIP - CAPI: Its ringing on the called party but Ive no sound. Ive tried codec ilbc and ulaw with no success. Does anyone have an idea? Thank you, Sincerely, Cyrille ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet
Here's the output of 'sip show users': *CLI sip show users UsernameSecret Accountcode Def.Context ACLNAT 502 1234 internalNo RFC35 501 1234 internalNo RFC35 If you need more info, just let me know. Time Bandit wrote: I have 2 Gnet SIP phones connected on the same switch as the Asterisk box. So far, our phones authenticate with *, because when I do sip show users, I see our 2 phones there. When you say that you see them, does it look something like this : 501/501172.16.1.201 D 255.255.255.255 5060 Unmonitored or like this : 501/501(Unspecified) D 255.255.255.255 5060 Unmonitored If it is (Unspecified) then the phone are not registering. The problem I have is this, when I try to dial the other extension, in this case 502, from 501, after a few seconds, I get a busy signal. If I check on the phone's logs, it says connection timeout. Do you have the output from Asterisk's CLI ? that would help us help you From a quick glance at your config, everything seems fine. B.T.W. I'm near you as I live in Brossard hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jean-Francois Theroux Systems administrator PrivalODC 450.761.9973 http://www.privalodc.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet
Jean-Francois Theroux a écrit : Here's the output of 'sip show users': *CLI sip show users UsernameSecret Accountcode Def.Context ACLNAT 502 1234 internalNo RFC35 501 1234 internalNo RFC35 If you need more info, just let me know. Your phones are described with host=ip address so no need of secret. Or you remove secret in sip.conf or you put your host as dynamic. And you setup accordly your phones. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newer Dell Servers + TDM card
Has anyone ever been able to fix this NMI power issue that the Dell's have with the TDM cards? Basically locks the machine up when trying to bring up the module. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet
Jean-Francois Theroux wrote: Here's the output of 'sip show users': *CLI sip show users UsernameSecret Accountcode Def.Context ACLNAT 502 1234 internalNo RFC35 501 1234 internalNo RFC35 sip show peers is the command you are looking for. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX aproprietary protocol
On Thu, 2005-04-28 at 10:33 +0400, Jean-Michel Hiver wrote: Joseph wrote: Can anybody explain me why IAX is called proprietary protocol? In some places IAX is refereed as open protocol. How can proprietary protocol be open protocol? Since the source code is available to anyone and GPL'ed it is an open protocol. However it's not a standard and there is no clear specification other than the source code (which kind of sucks because it means that ATA makers probably won't support it). Not really, I've exchanged few emails with one of the ATA manufacture in Taiwan and was able to convince them to add IAX2 to their ATA units; so by Q3 they said they will add it. I think is is the matter of speaking out and let them know what we are interested in (no IAX2 no sale). Once one or two of them will implement IAX2 in their units it will be like a chain reaction soon most of them will be supporting this protocol. ATCOM manufacture of AG-units in China will be adding support for IAX2 (according to their FAQ on their web-page). Soundwin in Taiwan will be adding it a well. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connection Timeout problem with SIP phones from Gnet
Ok. That's different. Here's the output of 'sip show peers': *CLI sip show peers Name/username Host Dyn Nat ACL Mask Port Status 502 172.16.1.202 255.255.255.255 5060 Unmonitored 501 172.16.1.201 255.255.255.255 5060 Unmonitored Eric Wieling aka ManxPower wrote: Jean-Francois Theroux wrote: Here's the output of 'sip show users': *CLI sip show users UsernameSecret Accountcode Def.Context ACLNAT 502 1234 internalNo RFC35 501 1234 internalNo RFC35 sip show peers is the command you are looking for. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jean-Francois Theroux Systems administrator PrivalODC 450.761.9973 http://www.privalodc.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Agents
I wanted to know if there is some way i can restrict the number of agents logged into one SIP extension. I usually find 2 or 3 agents logged on to a single extension. Can someone help me in this regard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 800 number provider suggestions
Hi all, Can anybody recommend a good 1-800 number provider? Thanks Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RJ45 to RJ11?
pin reversal should not be an issue as the silicon labs chips on the TDM card handle tip ring in either case. I think you will find it is pin reversed. So flip the RJ45 Over Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Thursday, 28 April 2005 4:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RJ45 to RJ11? Connect the POTS pair to pins 4 and 5 in the RJ45, and you should be fine (I say this not having looked at the TDM400 specs, but from the perspective of standard wiring practice and the assumption that Mark et al followed same). Greg Paul Shiflet wrote: I just received my TDM400 card from digium with 2 fxo and 2 fxs interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS phones. How do i interface my POTS phones with this; can i just crimp an RJ45 connection on the end of the phone cord? Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 25/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX aproprietary protocol
Stefan de Konink wrote: On Wed, 27 Apr 2005, Joseph wrote: How can proprietary protocol be open protocol? If the protocol is fully documentated and this documententation is available to anyone you can speak of a open protocol. It is not an open 'standard', because it is only supported by Digium, thus proprietary. http://en.wikipedia.org/wiki/Proprietary But there are royalties or something like that ? I understand that proprietary protocols CAN be published, but what make them proprietary is the requiremenf or royalties or at least a 'ok' from the owner ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to use dialparties.agi
Hi!
Thank you for answering my mail.
I think my mail wasn't exactly enough.
I want to use dialparties.agi in my own dialplan. I've tried it already,
but it don't work yet. I tried it the following way:
The users are dialing a number, this number is passed to a macro, which
calls dialparties.agi
exten = _XXX.,1,Macro(dial,60,tr,${EXTEN})
[macro-dial]
exten = s,1,AGI,dialparties.agi
exten = s,10,Dial(${ds}) ; dialparties
will set the priority to 10 if $ds is not null
exten = s,20,Wait(1) ;
dialparties will set priority to 22 if was direct call and caller is on
phone
exten = s,21,Voicemail(b${ARG3}) ; The call was internal to
extension, and was busy
This way the AGI is called and exited 0 but nothing happens...
Thanks,
Christian
P.S. I'm not sure if it is right to answer to the mailing list AND to
your mail address, please let me know if it was false.
David John Walsh wrote:
Christian
As I understand it
After a user dials an extension number, Asterisk calls dialparties.agi
dialparties.agi checks the asterisk database (show database [from
cli]) for data matching items like Call Wating (CW) Call Forward (CF)
etc.
If one is present (in a defiend order) then rather than dialing,
dialparties invokes that option.
If none of the options are set, dialparties returns control back to a
near regular dial string, and Dial takes over and places the call as
the A party was expecting.
Using defined etensions (by default in AMP they are the regular
American ones), the B party (callee) can activate these features.
What basically happens here is a database put command is used to put
the value in the asterisk database and then play a recorded
anouncement to the user before hanging the call up. for CF its a
little more complicated as you might have to specify the B number and
the C number, but essentially it puts the data in the database and
confirms it
Now the only thing that is missing is a web / gui provsioning system -
so that admins can take the features off again, else its a databse
del command at the terminal
---
the best way to see this in action is to set some things like CW (*73
i think) and then do a show database at the CLI - you will also get
back other things like the SIP registery
David
On 4/26/05, Christian Wengel [EMAIL PROTECTED] wrote:
Hi!
I looking for an example how to use the dialparties.agi from Asterisk
Management Portal 1.10.007a.
I tried to understand it by reading the extensions.conf of AMP, but
without success.
Is anybody out there, who can give me a more easy example or an explanation.
Thanks,
Christian
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[Asterisk-Users] Advice on Adtran 600 setup
I have a used Adtran 600 with 2 X Quad FXS, 2 X Dual FXS/DualX.3 and 3 X Dual FXO/Dual X3 modules. I have a Sangoma A101 Kit with RJ45 cable. Installed in my working Asterisk system is a Digium TDM22B 2 x FXS,2XFXO card. I would like to replace the Digium card with the Adtran unit, can anyone give me advice on configuration, I have no experience with the Adtran unit. Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (646)722-0001 Fax: (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents CallBackLogin and HangUp to calling party on pick-up
Hello, we have setup a queue with a couple of agents, all of which are joining in via CallbackLogin. 1 out of 10 calls coming into the queue will get hung-up upon as soon as the agent picks up the phone. We are running 1.0.6 bristuff RC7k (single HFC-card). SIP phones, ATAs and outside mobile phones. Anyone else experience this kind of behaviour? Anything I can do to pinpoint this problem? Thanks for any pointers. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and CISCO 7960?
Is someone running mgcp firmware with asterisk? I need to verify the phone issues Thanks. Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 FAX
my problem is i need to configure H323/Fax in asterisk to catch H323/Fax from the gateway and route it as t38/fax to another pbx server i installed on windows. how can i configure, route and convert the faxes? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime voicemail
Thank you both for the insight. The original problem was that the voice
mail system returned a no mailbox found error since the query was looking
for a mailbox in the default context and I had defined them in other
contexts, in my case, from-sip and analog-phones. It seems I am
confusing extension context with voicemail context. I included the
following in my extension file:
exten = 2201,1,agi,notify.agi
exten = 2201,2,Dial(Zap/9,20)
exten = 2201,3,Answer
exten = 2201,4,Wait(1)
exten = 2201,5,Voicemail(u${EXTEN})
exten = 2201,6,Hangup
exten = 2201,105,Voicemail(b${EXTEN})
exten = 2201,106,Hangup
For the channel definition in the zapata.conf file, I have the following:
context = analog-phones
group = 3
pickupgroup = 3
signalling = fxo_ks
adsi = yes
mailbox = [EMAIL PROTECTED]
callerid = Phone 1 2201
channel = 9
I realize that I did not need to use the EXTEN variable, since I had unique
entries in this case. I added [EMAIL PROTECTED] ( or could have used
the variable) and all works correctly. Thank you. I assumed that the
context entry in the voicemail_users table identified the mailbox
location. In the past, before realtime, and with the mailboxes defined in
voicemail.conf, I did not have to append the context in the extension table.
I don't really care that it is required now, but why did it work before?
Regards,
Ed Horton
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Re: [Asterisk-Users] Linksys/Cisco buys Sipura
Also, shortly after cisco purchased linksys (and with a little journalistic help), many of the past problems with their software finally 'surfaced' and were addressed. Given cisco's long history (internally) to standards and quality, I'd have to take some sizable bets on product improvement as opposed to anything else. I think its a win win situation. Cisco has tons of money to throw at them to get a better product with more features. I dont believe they would aquire them and not put money in them to make a better product. I guess the prices will go up like a rocket Not necessarily, When Cisco acquired linksys the prices of the linksys equipment went down. Guess you never know until it happens. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-h.323
Hi I've a problem with the registration of the openh323gatekeeper. First I've downloaded and installed the pwlib and openh323 libraries successfully. Then I've downloaded the package openh323gk.tar.gz,executed the binary file, but the gatekeeper is not registred on asterisk! Then I've also downloaded and installed the pwlib and openh323 over the Asterisk's pc, and launch the make command in the directory /../asterisk/channels/h323, as suggested by README file, in order to compile h323. I've several compilation errors related on ast_h323.o. Can someone help me about it?Are the installation steps correct? Thanks for all ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 doesn't know the hangup signal in china
I have a TDM400 with 4 fxo ports installed in my IPPBX box. When I call in my IPPBX through this card and after it answers I hangup, IPPBX still keeps going to timeout. It cannot recognize the hangup signal from PSTN. Anyone knows the solution. Past problems with disconnect generally fall into catagories of understanding 'exactly' what type of disconnect supervision is truly provided by the central office (if any), and setting the options to support it. Since you didn't provide any clues as to where you're located or what you've done to identify 'disconnect supervision', I'd suggest doing a little research on the wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX aproprietary protocol
Julio Arruda wrote: But there are royalties or something like that ? I understand that proprietary protocols CAN be published, but what make them proprietary is the requiremenf or royalties or at least a 'ok' from the owner ? Obviously IAX/IAX2 does not and should not require licensing fees. I once suggested to Mark that he copyright and trademark the terms IAX and IAX2 and provide a no-cost license for use of the term IAX or IAX2 to any implimentation that correctly follows the (at some point) documented IAX2 protocol. He seemed to like the idea, but I don't know if anything was actually done. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
In your case, where you will need the license is on the box that your phones register to. For exampe, when someone checks voicemail, encoding takes place, therefore you need a license. Look at it this way: [g729 provider] -(SIP or IAX)--- [g729 asterisk server] - no license required in the above connection if using g729 solely [g729 asterisk server containing non-g729 audio files] (SIP)- [g729 SIP Phone] - a license is required above for each non-g729 audio file or stream that needs to be encoded to be sent out as g729 to the g729 SIP Phone (ie. voicemail, IVR prompts, etc.). Hope that makes sense. On 4/28/05, Matt [EMAIL PROTECTED] wrote: For instance.. when I try to use G723.1 on my phone (and just call in from my PRI line) I get: Unable to find a path from g723 to ulaw. Unable to find a path from ulaw to g723. No path to translate from Zap/1-1(68) to Sip/201-80c7(1). Same things happens if I call in on my current provider's number which uses G711 for the codec. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960s and skinny
Paul wrote: Do you still have that image for the 7960? I bought a 7940 on ebay and it doesn't have the SIP firmware. I can't find it anywhere but Cisco's website and they require that I have an account with them. Did you happen to save that binary file? Cisco charges for the SIP firmware. You can purchase it via an authorized Cisco reseller. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Firmware Price Increase
Not that I know of I am a Cisco partner and the Category 1 contract is still at least half that or less. He was talking about the SIP-license... Not the SmartNET. If you have a SmartNET, you CAN download the SIP load but to use it, you need the license. I think that's the point; to use sip please pay an additional $150US. Downloading the image is supposedly illegal unless you have a license. Now, what is the true list price of a new 7960 with sip? (Be careful to read the license terms before answering that question.) You suffer from two fundamental misunderstandings: I don't think so, but maybe we're bumping up against the words that are being used here. Split this into three questions, and answer those questions with some assurance that legal licensing issues are addressed. 1) part of the discussion was if you owned a 7960, what is required to obtain a sip image and legally use it (that obviously assumes some other image is installed on the 7960)? 2) if one ordered a brand new 7960 from an authorized cisco reseller, what exactly 'must' be purchased (with costs) to use that 7960 in a sip environment? 3) if one purchased a 7960 on ebay (or some non-cisco reseller), what exactly must be ordered to legally use that phone in a sip environment? 1) The SmartNET does not permit you to download the firmware. It does permit you to UPGRADE your licensed firmware. It might be Cisco's error to have not separated the downloads by firmware versions or protocols. However, to be able to do something never meant to be allowed to do it, right? 2) You also need a license for using a 7960 with the Cisco Call Manager. The trick is to figure out if you bought the license with the CCM itself or if you need to buy an additional CCM One Station User License which cost exactly the same as the SIP license. If you want a SIP phone and know it before buying, you should buy a phone WITHOUT the CCM license. Most dealers sell Cisco IP phones INCLUDING this CCM license. To be sure, order a spare phone: CP-7960G= Cisco IP Phone 7960G, Global, Spare D $415 CP-7940G= Cisco IP Phone 7940G, Global, Spare D $315 CP-7936= IP Conference Station, SpareD $1.195 CP-7912G= Cisco IP Phone 7912G, Global, Spare D $245 CP-7905G= Cisco IP Phone 7905G, Global, Spare D $165 Don't forget the power supply if you don't have a Cisco PoE switch CP-PWR-CUBE-2 IP Phone power transformer for the 7900 phone series D $45 and the power cord CP-PWR-CORD-NA= 7900 Series Transformer Power Cord, North America N/A $10 CP-PWR-CORD-CE= 7900 Series Transformer Power Cord, Central Europe N/A $10 CP-PWR-CORD-UK= 7900 Series Transformer Power Cord, United Kingdom N/A $10 CP-PWR-CORD-AU= 7900 Series Transformer Power Cord, Australia N/A $10 CP-PWR-CORD-JP= 7900 Series Transformer Power Cord, Japan N/A $10 CP-PWR-CORD-AP= 7900 Series Asia Pacific Power CordN/A $10 Not to mention that street prices are substantially lower than Global Pricelist values... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-h.323
Hi, What do you have h323 or oh323 , (open h 323) I think you have the latest. You must use the SPECIFIC files. Check http://www.inaccessnetworks.com/projects/asterisk-oh323 Also PATCH the file BEFORE compilation It should run then. Good luck Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: jeudi 28 avril 2005 16:05 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk-h.323 Hi I've a problem with the registration of the openh323gatekeeper. First I've downloaded and installed the pwlib and openh323 libraries successfully. Then I've downloaded the package openh323gk.tar.gz,executed the binary file, but the gatekeeper is not registred on asterisk! Then I've also downloaded and installed the pwlib and openh323 over the Asterisk's pc, and launch the make command in the directory /../asterisk/channels/h323, as suggested by README file, in order to compile h323. I've several compilation errors related on ast_h323.o. Can someone help me about it?Are the installation steps correct? Thanks for all ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
So [g729 provider] -(SIP or IAX)--- [g729 asterisk server] This is how I'd be setup.. actually more like this: [g729 provider] --(sip) [g729 asterisk server](sip)---[g711 sip phone client]. So... if I understand this correctly.. I *would not* for *any* reason need a license going from the g711 client since to voicemail/etc is fine.. and going out to the provider is not in my system? However, if someone calls IN from the g729 provider and wants to check voicemail, then I'd need a license? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] proper 2-card ISDN modem.conf configuration?
I'm trying to configure an asterisk box with two cards.
Incoming calls are working fine with two ISDN cards, however, I am able
to make outgoing calls only through the first card.
exten = _0.,1,Dial(Modem/g1:${EXTEN:1})
exten = _9.,1,Dial(Modem/g2:${EXTEN:1})
If I try to use the second card, asterisk says that the line is busy
(which isn't true).
So I thought that maybe my modem.conf is wrong?
Could you paste your modem.conf here, if you are using more than one
ISDN card?
Below my modem.conf:
[interfaces]
context=remote
driver=i4l
language=de
type=i4l
dialtype=tone
mode=immediate
dtmfmode=both
group=1
msn=27229933
incomingmsn=*
device = /dev/ttyI0
device = /dev/ttyI1
group=2
msn=624
incomingmsn=*
device = /dev/ttyI2
device = /dev/ttyI3
Tomek
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[Asterisk-Users] Prefix to CALLING Number ?
Hi there, I`m trying to addsome prefixbefore my local extensions, when my calls are routed to ZAP trunk. (i.e.: my local extension is , and i would like to send request to my telco provider with source phone number 55) Is there any way to do this ? I just know toadd prefix (via prefix application) to the called number (but not calling). Thanks, barney PS: sorry for my poor english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
On Thu, 2005-04-28 at 11:03 -0400, Matt wrote: So [g729 provider] -(SIP or IAX)--- [g729 asterisk server] This is how I'd be setup.. actually more like this: [g729 provider] --(sip) [g729 asterisk server](sip)---[g711 sip phone client]. So... if I understand this correctly.. I *would not* for *any* reason need a license going from the g711 client since to voicemail/etc is fine.. and going out to the provider is not in my system? No, you will ALWAYS need a license while any call is active between your provider and your asterisk box. The asterisk box (in your description) needs to convert from G.729 to G.711, therefore, it needs a license to do that. Pay $10, get a license, and move on, it will make your life significantly easier. Or, find a provider that will do gsm, or G.711. Or, use a phone that has G.729, and don't use any asterisk functions such as voicemail/etc... -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL
I searched for the mailing list guidelines on google and couldn't find them. I apologize in advance if this is not the appropriate list. My company is moving their office and we have decided to use VoIP for our phone solution. We will be using Cisco 7960 phones powered by a Cisco 3560 switch. The server running Asterisk will be a Dell 2650 Dual Xeon with 2GB of RAM running Linux. We need to set this system up in the next month and I was wondering if there are any Asterisk consultants in the Chicagoland area to assist us in the initial setup and quite possibly on an as needed basis? We are located in the Loop area. Regards, Jon Dahl _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime voicemail
exten = 2201,1,agi,notify.agi
exten = 2201,2,Dial(Zap/9,20)
exten = 2201,3,Answer
exten = 2201,4,Wait(1)
exten = 2201,5,Voicemail(u${EXTEN})
exten = 2201,6,Hangup
exten = 2201,105,Voicemail(b${EXTEN})
exten = 2201,106,Hangup
I realize that I did not need to use the EXTEN variable, since I had
unique entries in this case. I added [EMAIL PROTECTED] ( or
could have used the variable) and all works correctly. Thank you. I
assumed that the context entry in the voicemail_users table
identified the mailbox location. In the past, before realtime, and
with the mailboxes defined in voicemail.conf, I did not have to
append the context in the extension table. I don't really care that
it is required now, but why did it work before?
Don't know why it worked before. I've always had Voicemail([EMAIL PROTECTED]) in
all my extensions from day 1.
-Matthew
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Re: [Asterisk-Users] Confused on G723 and G729
I'll gladly pay $10 a license... I'm all for supporting digium... however, I was under the impression that there was also some huge one time fee of like $2,000 or something. I guess I was wrong... ok now bad.. So I purchase the license from digium... then what happens/what needs to be done on Asterisk? On 4/28/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Thu, 2005-04-28 at 11:03 -0400, Matt wrote: So [g729 provider] -(SIP or IAX)--- [g729 asterisk server] This is how I'd be setup.. actually more like this: [g729 provider] --(sip) [g729 asterisk server](sip)---[g711 sip phone client]. So... if I understand this correctly.. I *would not* for *any* reason need a license going from the g711 client since to voicemail/etc is fine.. and going out to the provider is not in my system? No, you will ALWAYS need a license while any call is active between your provider and your asterisk box. The asterisk box (in your description) needs to convert from G.729 to G.711, therefore, it needs a license to do that. Pay $10, get a license, and move on, it will make your life significantly easier. Or, find a provider that will do gsm, or G.711. Or, use a phone that has G.729, and don't use any asterisk functions such as voicemail/etc... -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL
We do a good amount of remote work if that isn't a problem for you. We can reconfigure the entire system and have it ready to drop into place. If the job is big enough it might warrant a visit during installation but that isnt always the case. Kerry Garrison Tech Data Pros http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Dahl Sent: Thursday, April 28, 2005 8:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL I searched for the mailing list guidelines on google and couldn't find them. I apologize in advance if this is not the appropriate list. My company is moving their office and we have decided to use VoIP for our phone solution. We will be using Cisco 7960 phones powered by a Cisco 3560 switch. The server running Asterisk will be a Dell 2650 Dual Xeon with 2GB of RAM running Linux. We need to set this system up in the next month and I was wondering if there are any Asterisk consultants in the Chicagoland area to assist us in the initial setup and quite possibly on an as needed basis? We are located in the Loop area. Regards, Jon Dahl _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX attempt - Segmentation fault
Hello,
I can't use IAX with my last
CVS-NHEAD-04/28/05-16:00:04 installation.Every time I try to use an iax
channel or register an iax user, I get a Segmentation fault.
Trace:
-- Executing Dial("SIP/25-0368",
"IAX2/25|20|Tt") Segmentation fault
[EMAIL PROTECTED] root]# Ouch ... error while writing audio data: : Broken
pipe Warning, flexibel rate not heavily
tested!
Regards,
Victor.
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[Asterisk-Users] Console Warning Message
Does anyone know what this mean? Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Received mini frame before first full voice frame Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue member persistent stats
Matthew Boehm wrote: I've got 5 agents who login/logff via AddQueueMember. Each time they do so, their stats get reset. Is there anyway to keep these stats across logins? Nobody has implemented that to date that I am aware of. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] start asterisk
Hi All. I have installed Asterisk on linux Redhat version 9, I follow step by ssstep the installation, my card digium is TDM400P, whith modprobe wcfxs I have load this module. My vonfiguration files are in /etc/asterisk, the file /etc/zaptel.conf hace the folloeing lines: fxoks=1 #fxsks=4 loadzone=us defaultzone=us whit command ztcfg -vv say: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. Nut when I start the asterisk the following message appear Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: manager.c:1478 in init_manager: Unable to open management configuration manager.conf. Call management disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_agent.c:809 in read_agent_config: No agent configuration found -- agent support disabled Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_mgcp.c:3948 in reload_config: Unable to load config mgcp.conf, MGCP disabled Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_iax2.c:6839 in set_config: Unable to load config iax.conf Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: iax2-provision.c:496 in iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_skinny.c:2541 in reload_config: Unable to load config skinny.conf, Skinny disabled Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:1016 in load_module: XXX I don't work right with non-full duplex sound cards XXX Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:257 in sound_thread: Read error on sound device: Resource temporarily unavailable Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:6220 in mkintf: Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:9155 in setup_zap: Unable to register channel '1' Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:345 in ast_load_resource: chan_zap.so: load_module failed, returning -1 Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:440 in load_modules: Loading module chan_zap.so failed! Somebody can give me suggestions? Thanks in Advanced, Regards. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues configuration
Anton Krall wrote: How do you do it? I mean, if a caller is already on the queue and suddenly all agents logoff.. How do you make the caller fall out of the queue and into an IVR where he can leave a message? Have you read the sample queues.conf file? There is an option there called 'leavewhenempty' that does exactly that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Number of production asterisk systems
Hey Guys, I lost a deal to another vendor because he could point to a number of installations of the product he was selling in our area and nationally, even though _he_ didn't implement them directly. Very frustrating, but I don't imagine uncommon as we compete with other more recognizable solutions. What I am trying to do is track down a rough idea of how many Asterisk systems are in production right now. Ideally as this information was gathered it could be sorted by country, state, industry, etc. Does anyone have any information, or any idea of where to start? Any of the Digium guys on this list know if Digium attempts to track this sort of information? I would be willing to donate time to help compile and correlate the information if anyone has any idea where to start. In the end, I think it would benefit the entire community. I considered posting this to the -biz list but in the end decided that since I was not look for or offering services, goods, etc. that the -user list was a better place. I apologize if you disagree. I know this list is bursting at the seams already. ~chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] start asterisk
did you do the following service zaptel stop service zaptel start then run asterisk... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX attempt - Segmentation fault
Victor Alvarez wrote: Hello, I can't use IAX with my last CVS-NHEAD-04/28/05-16:00:04 installation. Every time I try to use an iax channel or register an iax user, I get a Segmentation fault. Trace: -- Executing Dial(SIP/25-0368, IAX2/25|20|Tt) Segmentation fault [EMAIL PROTECTED] root]# Ouch ... error while writing audio data: : Broken pipe Warning, flexibel rate not heavily tested! This is mpg123 error, not an IAX2 error. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] start asterisk
You need to change the line type in either your zapata.conf or your zaptel.conf they need to match. - Original Message - From: Luz Lopez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, April 28, 2005 11:01 AM Subject: [Asterisk-Users] start asterisk Hi All. I have installed Asterisk on linux Redhat version 9, I follow step by ssstep the installation, my card digium is TDM400P, whith modprobe wcfxs I have load this module. My vonfiguration files are in /etc/asterisk, the file /etc/zaptel.conf hace the folloeing lines: fxoks=1 #fxsks=4 loadzone=us defaultzone=us whit command ztcfg -vv say: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. Nut when I start the asterisk the following message appear Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: manager.c:1478 in init_manager: Unable to open management configuration manager.conf. Call management disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_agent.c:809 in read_agent_config: No agent configuration found -- agent support disabled Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_mgcp.c:3948 in reload_config: Unable to load config mgcp.conf, MGCP disabled Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_iax2.c:6839 in set_config: Unable to load config iax.conf Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: iax2-provision.c:496 in iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_skinny.c:2541 in reload_config: Unable to load config skinny.conf, Skinny disabled Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:1016 in load_module: XXX I don't work right with non-full duplex sound cards XXX Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:257 in sound_thread: Read error on sound device: Resource temporarily unavailable Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:6220 in mkintf: Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:9155 in setup_zap: Unable to register channel '1' Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:345 in ast_load_resource: chan_zap.so: load_module failed, returning -1 Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:440 in load_modules: Loading module chan_zap.so failed! Somebody can give me suggestions? Thanks in Advanced, Regards. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delete voicemail
Hi all, Does anyone know what the easiest way is to delete voicemail for one extension? Had a search online but couldn't find anything. Cheers, Damian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] start asterisk
On Thu, 28 Apr 2005 16:01:44 + Luz Lopez [EMAIL PROTECTED] wrote: Hi All. I have installed Asterisk on linux Redhat version 9, I follow step by ssstep the installation, my card digium is TDM400P, whith modprobe wcfxs I have load this module. My vonfiguration files are in /etc/asterisk, the file /etc/zaptel.conf hace the folloeing lines: fxoks=1 #fxsks=4 loadzone=us defaultzone=us whit command ztcfg -vv say: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. Nut when I start the asterisk the following message appear Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: manager.c:1478 in init_manager: Unable to open management configuration manager.conf. Call management disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_agent.c:809 in read_agent_config: No agent configuration found -- agent support disabled Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_mgcp.c:3948 in reload_config: Unable to load config mgcp.conf, MGCP disabled Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_iax2.c:6839 in set_config: Unable to load config iax.conf Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: iax2-provision.c:496 in iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_skinny.c:2541 in reload_config: Unable to load config skinny.conf, Skinny disabled Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:1016 in load_module: XXX I don't work right with non-full duplex sound cards XXX Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:257 in sound_thread: Read error on sound device: Resource temporarily unavailable Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:6220 in mkintf: Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:9155 in setup_zap: Unable to register channel '1' Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:345 in ast_load_resource: chan_zap.so: load_module failed, returning -1 Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:440 in load_modules: Loading module chan_zap.so failed! Somebody can give me suggestions? Thanks in Advanced, Regards. Did you put the correct settings in zapata.conf as per the wiki?? http://www.voip-info.org/wiki-Asterisk+config+zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer
I just bought one of these zyxel wireless phones, of course there is no transfer key. Is there a patch for the stable 1.0.7 that I can implement # or any other key or combination to initiate a transfer? I looked briefly through the wiki and archived lists and didn't see much. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Delete voicemail
Command line on the box and navigate to the directory for your VM. An example of one of mine... /var/spool/asterisk/voicemail/default/1003/INBOX/ Issue the rm *.* command Bye bye files Your location may vary slightly depending on what * you are using. I am on AAH 0.9. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damian Funnell Sent: Thursday, April 28, 2005 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Delete voicemail Hi all, Does anyone know what the easiest way is to delete voicemail for one extension? Had a search online but couldn't find anything. Cheers, Damian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Confused on G723 and G729
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, April 28, 2005 8:31 AM To: Adam Goryachev Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Confused on G723 and G729 I'll gladly pay $10 a license... I'm all for supporting digium... however, I was under the impression that there was also some huge one time fee of like $2,000 or something. I guess I was wrong... ok now bad.. So I purchase the license from digium... then what happens/what needs to be done on Asterisk? Be aware that the license fee is $10 per instance. Each leg that is transcoded to or from G.729 on your box will use one license. So if you want to support 20 simultaneous callers checking their voicemail, you'll need 20 licenses. The installation process is well documented on Digium's web site. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.4 - Release Date: 04/27/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
You would need a transcoding license between the Asterisk PBX and the G711 phone... On 4/28/05, Matt [EMAIL PROTECTED] wrote: So [g729 provider] -(SIP or IAX)--- [g729 asterisk server] This is how I'd be setup.. actually more like this: [g729 provider] --(sip) [g729 asterisk server](sip)---[g711 sip phone client]. So... if I understand this correctly.. I *would not* for *any* reason need a license going from the g711 client since to voicemail/etc is fine.. and going out to the provider is not in my system? However, if someone calls IN from the g729 provider and wants to check voicemail, then I'd need a license? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BIND VoIP anyone?
Hi List, I was looking for, but I couldn't find any product or project like BIND that works with VoIP in an homologous way. I mean, is there anybody working in a way to register user-ids or domain name-like information so VoIP calls can be dialed in a number string format from any IP phone? Any clue? Thanks all, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-h.323
I believe that you have to start openh323gatekeeper before starting asterisk. Regards, Neal -Original Message- From: [EMAIL PROTECTED] [SMTP:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 7:05 AM To: asterisk-users@lists.digium.com Subject:[Asterisk-Users] asterisk-h.323 Hi I've a problem with the registration of the openh323gatekeeper. First I've downloaded and installed the pwlib and openh323 libraries successfully. Then I've downloaded the package openh323gk.tar.gz,executed the binary file, but the gatekeeper is not registred on asterisk! Then I've also downloaded and installed the pwlib and openh323 over the Asterisk's pc, and launch the make command in the directory /../asterisk/channels/h323, as suggested by README file, in order to compile h323. I've several compilation errors related on ast_h323.o. Can someone help me about it?Are the installation steps correct? Thanks for all ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 Technology and VoIP Gateway Primer
Asterisk Users / Asterisk Biz List Members, About a week ago I cross-posted a message titled Large Asterisk Setup (~500 Concurrent Calls + Scalability) to Asterisk-Users and Asterisk-Biz. For reference, the threads generated by that message are archived at the following locations: http://lists.digium.com/pipermail/asterisk-users/2005-April/102823.html http://lists.digium.com/pipermail/asterisk-biz/2005-April/004590.html First, I would like to thank you all for your excellent suggestions and contributions. My misunderstanding of the Digium quad-span card's scaling limitations was corrected (PCI bus traffic is not the problem, it is the number of interrupts generated by the Zaptel drivers) and I was directed to replace the Asterisk Slave Servers in this diagram (http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) with a VoIP gateway. Following up on that suggestion, I began researching T-1 time division multiplexing in order to understand where the DSP load on an Asterisk server originates and the best options for purchasing a VoIP gateway to offload that processing onto. The results of that research can be found at the following links: T-1 Multiplexing - PSTN Side (http://home.comcast.net/~mroth01/T1-PSTN.gif) T-1 Multiplexing - CPE Side (http://home.comcast.net/~mroth01/T1-CPE.gif) Basic T-1 Time Division Multiplexer (http://home.comcast.net/~mroth01/T1-TDM.gif) Telephony Glossary (http://home.comcast.net/~mroth01/telephony-glossary.html) Sources (http://home.comcast.net/~mroth01/sources.html) My understanding of the T-1 TDM and the PSTN side is pretty solid, as it is mainly based off of Intel Corporation's T1/E1 Technology Primer (see Sources), but the CPE side is largely deduced from what I knew about the PSTN side. There may be holes or mistakes, so I would appreciate any corrections or additions that you can offer. Specifically, I would like a detail of the TDM - VoIP conversion process, similar to the basic T-1 TDM one I provided. The differences between a T-1, DS-1, and ISDN are subtle and not universally agreed upon. For a discussion of these issues see the following links: What's the diff between a T1 and a DS1 (http://pbxtech.info/showthread.php?t=1100) PRI setup (http://pbxtech.info/showthread.php?t=1250) In closing, I have a few questions: - Is my understanding of using the same codecs and signaling protocols on both sides of the Asterisk server in order to circumvent transcoding and conversions on the server correct? - Are there any other host-intensive processes that I should consider offloading to the gateway, such as echo cancellation? - What does the PCM µ-law codec used in T-1 multiplexing map to in terms of Asterisk codecs (G.711 µ-law, perhaps)? - What codec does the Monitor application use when digitally recording calls (if possible, I would like to avoid transcoding the streams when recording and let sox handle the conversions on a different box)? Thank you for your time, Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP calling Error from MP108 please help - confs included
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
___
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Re: [Asterisk-Users] Redirect two channels to each other?
It almost sounds like there needs to me a new manager action: Action: Bridge ChannelA: SIP/199testfone-1f3c ChannelB: Zap/6-1 It sounds like the intrinsic functionality for 'bridging' is already there in Asterisk (duh!), it just needs to be encapsulated in a manager action. Yes, we need that action on the manager! (but replace ChannelA and ChannelB to Channel1 and Channel2 as on the link event). -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIND VoIP anyone?
Take a look to ENUM http://www.enum.org/ - Original Message - From: Andres Paglayan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 28, 2005 6:39 PM Subject: [Asterisk-Users] BIND VoIP anyone? Hi List, I was looking for, but I couldn't find any product or project like BIND that works with VoIP in an homologous way. I mean, is there anybody working in a way to register user-ids or domain name-like information so VoIP calls can be dialed in a number string format from any IP phone? Any clue? Thanks all, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk call generator
Hi all Am looking for a way to generate like 300 simultanious calls to test *'s perfomance on a big load. * is currently working perfectly with H323, sip and IAX. Any suggestions are welcome Sam Njenga ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] start asterisk
Hi, I have installed zaptel, but I haven't in /etc/rc.d/init.d the file to start zaptel. From: Jerry Geis [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] start asterisk Date: Thu, 28 Apr 2005 11:08:17 -0500 did you do the following service zaptel stop service zaptel start then run asterisk... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicpulseConnect problems?
Folks, I'm having trouble with my voicepulse numbers. Over the past week, incoming calls have been very slow to be answered, but they seem fine while the call is in progress. When the caller hangs up, asterisk takes a while (over 2 minutes in some cases). This system does not make outgoing calls. Today, after rebooting my machine and rotating the log files, I have absolutely NO incoming calls being received. My cell phone dials the number, tells me it's connected, and then happily hangs up 10 to 12 seconds later, while asterisk (and the logs) show no indication at all of any incoming calls. Looking at my syslog and asterisk messages, the only thing I'm seeing over the past week that did not use to happen is this message in the asterisk logs: Apr 28 10:06:45 WARNING[4282]: Host 'gwiax-in-01.voicepulse.com' not found at line 72 But that's been happening for about the same time as the slow-down issue, and still calls _were_ being answered, albeit slowly. I'm HoSed. :) Has anyone else run into this? Got any ideas on what's up at VPConnect? Do I need to placate the rain-god or something? Any help would be appreciated! Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP calling Error from MP108 please help - confs included
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
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Re: [Asterisk-Users] ATA 186 MGCP Firmware
I've been there.. the page comes up with There are currently no files for this type. Well, you either have a technical problem or an administrative one. Eliminate the possibility of corrupted cookies or browser cache by going to another workstation, accessing http://www.cisco.com/cgi-bin/tablebuild.pl/ata186 and entering your credentials. If you still see no files listed, it appears that Cisco has (perhaps inadvertently) downgraded your account. Open a case with them. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BIND VoIP anyone?
Don't forget Dundi is such a system that is already integrated into Asterisk. http://www.dundi.info/ Original Message Subject: [Asterisk-Users] BIND VoIP anyone? From: Andres Paglayan [EMAIL PROTECTED] Date: Thu, April 28, 2005 11:39 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi List, I was looking for, but I couldn't find any product or project like BIND that works with VoIP in an homologous way. I mean, is there anybody working in a way to register user-ids or domain name-like information so VoIP calls can be dialed in a number string format from any IP phone? Any clue? Thanks all, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
Henry Devito wrote: I just bought one of these zyxel wireless phones, of course there is no transfer key. Is there a patch for the stable 1.0.7 that I can implement # or any other key or combination to initiate a transfer? I looked briefly through the wiki and archived lists and didn't see much. show application dial Pay special attention to the t/T options. Those options are for devices that are too stupid or brain dead to have a transfer key that works. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web interface Suggestions
Has anyone come across any software that can control adding/editing SIP extension properties and perhaps dial plan properties on a context basis. What I mean is I would like it so an admin user from Company A can manipulate properties for extensions in his context but not in another Companies. I know AMP does something similar to this but from what I understand it does not allow for different users at different companies to control only things that pertain to them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
