Re: [Asterisk-Users] SMP support
On Sat, 2004-09-25 at 09:49, Michael Bielicki wrote: 64bit it :) [EMAIL PROTECTED] root]# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 5 model name : AMD Opteron(tm) Processor 244 Any idea to the number of channels your system is capable of handling? I'm specifically interested in zap - sip channels which include transcoding to something like g.729 codec. In case anyone else has relative comments, I'm hoping to extract up to 240 simultaneous calls, although am looking at the 2 x Opteron 246 CPU's. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding inbound calls right back out
I have calls coming in via SIP (a DID) and I want to forward them right back
out to my cell.
If I do it in one step,
(as if 2125551212 was the DID, and 202111 was my cell number)
exten = 2125551212,1,Dial(SIP/${PROVIDER}/1202111,60)
The call comes in via sip, my system sends the invite for the outbound call,
the sends a cancel, and the caller hears ringing begin.
If I use
exten = 2125551212,1,Answer
exten = 2125551212,2,Dial(SIP/${PROVIDER}/1202111,60)
The outbound call happens, CLID is correct, can be answered, etc., but I
only get audio in one direction...from the phone that originated the call TO
the phone that answered the call.
I assume I've done something wrong?
Note that I can direct the inbound call to voicemail and it works fine, so
the inbound SIP session isn't likely the problem.
All help greatly appreciated!
Thanks,
Eric
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[Asterisk-Users] RE:[Asterisk-Dev] Free G.729 ready for download
I use Digium's Licensed Codec and I have no problems in routing calls to either E1 or T1 interfaces. But ...beware of the Pitfalls in using non-standard G729 Codecs. I used a couple of sets before and here are the problems I found (I have not used Daniels codec though): 1) Calls are too noisy and not at all readable 2) Calls disconnect as soon as connect 3) Calls required re-invite 4) Connectivity Timeouts 5) Authentication failures 6) E1 Gateways cannot decipher the calls I have gone back to my limited user Digium License from the free G729 and I could not be happier. I will give Daniels' codec a try if more people confirm (mainly those who have successfully ran it on Fedora Core2) that this hack really works. Seshu Kanuri Netweb Group, Inc. Ph:1-732-387-4133 Fx:1-413-812-3152 [EMAIL PROTECTED] www.netwebgroup.com This e-mail message may contain confidential, proprietary or legally privileged information. It should not be used by anyone who is not the original intended recipient. If you have erroneously received this message, please delete it immediately and notify the sender. The recipient acknowledges that Netweb Group, Inc. or its subsidiaries and associated companies, are unable to exercise control or ensure or guarantee the integrity of/over the contents of the information contained in e-mail transmissions and further acknowledges that any views expressed in this message are those of the individual sender and no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of Netweb Group, Inc. Before opening any attachments please check them for viruses and defects. - Original Message - From: Arkadi Shishlov [EMAIL PROTECTED] To: Asterisk Developers Mailing List [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Friday, September 24, 2004 1:10 PM Subject: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download I expropriated the right to rip Daniel's disclamer for use in my email too.. DISCLAIMER: You might have to pay royalty fees to the G.729 patent holders for using their algorithm. For easier testing I prepared codec_g729.so binaries and associated libraries and put them on the web: http://kvin.lv/pub/Linux/Asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
Hi, As I am the developer of DIAX - Original Message - From: Robert Rozman [EMAIL PROTECTED] there is already iax softphone called diax (http://www.laser.com/dante/diax/diax.html) that can be controlled over bluetooth on some phones. The thing that is missing is to be able to use cellular as audio device for softphone (I'm doing this with paired bluetooth headset - but that is not proper solution). We already have Audio gateway bluetooth profile that allows redirection from cellular to PCs sound card, but we would need same in opposite direction - to use cellular as PCs soundcard on softphone application. I cannot do it with my SonyEricsson T68i. If anyone can do it, then I'll integrate this feature in DIAX too. What I want to do first, but I don't know how is to control DIAX using the BT headset internal switch to answer the call. I don;t know how to start BT connection from the headset side:-( Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to get Call Details Records
Title: Message HI, Can anyone please tell me 1) Where does asterisk store the call detail records? 2) What is thestructure of these call details records? 2)How to access the call detail records by any external application? Thanks in advance Regards, Mayank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian Sarge, ISDN, CAPI and Asterisk blues
Hi all, I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest testing versions). If I start asterisk I get: chan_capi.c:2635 load_module: CAPI not installed. lsmod | grep capi gives: capi17472 0 capifs 60242 capi kernelcapi 46496 1 capi Anyone any suggestions of where to look? Anyone a working asterisk with ISDN on Debian? Groeten, Joost Kraaijeveld Askesis B.V. Molukkenstraat 14 6524NB Nijmegen tel: 024-3888063 / 06-51855277 fax: 024-3608416 e-mail: [EMAIL PROTECTED] web: www.askesis.nl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
On Sat, 25 Sep 2004, Steve Underwood wrote: On Sat, 25 Sep 2004, Steve Underwood wrote: I am not a lawyer, nor even a US citizen. Talking to someone who is both may be a good idea. What is the relevance of being a US citizen? Copyright rules are largely global. There are two different major sets of copyright laws, depending on which treaty they were derived from. They are not always compatible. They differ in such points as whether you can transfer your copyright or merely assign the rights granted by it. Unless otherwise granted by the copyright holder, by default the copyright of a derived work (in the copyright legan sense) is held by the owner of the original copyright and not the crator of the derived work. So no, the patches are owned by Intel as well. But the patches aren't a derived work. That is the value they have here. There are an independant adjunct work. According to most lawyers a patch _is_ a derived work in nearly all circumstances. E.g. a novel based on the characters from a novel by another author is a derived work. If you are producing copies of just about anything you really need to speak to your lawyer to be safe. The excpetion possibly being open source stuff based soley on open source stuff. Anyway, this is getting too far off topic for this list. Mea culpa. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
find someone to host it in India or serbia and you can safely ignore it :) On Sat, 25 Sep 2004 10:22:52 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 25 Sep 2004, Steve Underwood wrote: On Sat, 25 Sep 2004, Steve Underwood wrote: I am not a lawyer, nor even a US citizen. Talking to someone who is both may be a good idea. What is the relevance of being a US citizen? Copyright rules are largely global. There are two different major sets of copyright laws, depending on which treaty they were derived from. They are not always compatible. They differ in such points as whether you can transfer your copyright or merely assign the rights granted by it. Unless otherwise granted by the copyright holder, by default the copyright of a derived work (in the copyright legan sense) is held by the owner of the original copyright and not the crator of the derived work. So no, the patches are owned by Intel as well. But the patches aren't a derived work. That is the value they have here. There are an independant adjunct work. According to most lawyers a patch _is_ a derived work in nearly all circumstances. E.g. a novel based on the characters from a novel by another author is a derived work. If you are producing copies of just about anything you really need to speak to your lawyer to be safe. The excpetion possibly being open source stuff based soley on open source stuff. Anyway, this is getting too far off topic for this list. Mea culpa. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ISDN (point to point) questions
Hello Bjoern, thanks for this nice discussion; we we dod have msn (4) although the telco company tells us that we have pp isdn. This seems to be a little bit strange to me... Is there any way to crosscheck the isdn configuration ? And what about the active or passive isdn cards ? I just want to drive voice and fax over them. -- Best regards, Dannymailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. Saturday, September 25, 2004, 12:47:41 AM, you wrote: BA Hi Danny, BA dont mix ppp and p2p... BA pp = point to point BA pmp = point to multipoint BA (both are ISDN connection configurations) BA ppp = point to point protocol BA a higher level protocol for data transmission etc. BA Of which type your ISDN connection is, is usually easy to decide looking BA at your phonenumbers BA A PMP has MSNs (usually (in Germany) up to 10 with EuroISDN) BA There may be up to (I'm not sure...) 8 devices connected to one PMP BA ISDN connection. BA A PP has one base number with DIDs (like 234567-0 234567-10 234567-11 BA etc). There may only be one device connected to the ISDN connection. BA And with PP you can have several BRIs or PRIs 'sharing' the same number. BA Greetings BA Bjoern BA Danny Zak schrieb: Hello; we are looking to replace our current PBX with a *-box; it is connected to ONE ppp isdn connection that is terminated by the NC. We got on this box 4 msn's configured. currently we are working with pstn fxo's behind the PBX; it works but we can't use the CSID information behind it. We want to migrate and keep the MSN's to decide routing in combination with the CID. That's why we want to replace all our phones (8) with voip ones (or by using a fxs's) Reading all information i assume the following - we need a ACTIVE ISDN card; these are 5/6 times more expensif that passive ones. I always assumed there was only PPP and PMP; but it seems - reading the specs of the av. isdn card - that there is also multilink PPP. Will a multilink PPP also support a PPP; or is this just a other way to describe a PMP ? - how can i checke the number that is being dialed by the caller to reachh the * box (so one of the 4 msn's). I have seen dialplans making use of the CALLERIDNUM; but what do i need to query for the called num ? BA ___ BA Asterisk-Users mailing list BA [EMAIL PROTECTED] BA http://lists.digium.com/mailman/listinfo/asterisk-users BA To UNSUBSCRIBE or update options visit: BAhttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Sarge, ISDN, CAPI and Asterisk blues
On Sat, Sep 25, 2004 at 10:12:44AM +0200, Joost Kraaijeveld wrote: Hi all, I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest testing versions). If I start asterisk I get: chan_capi.c:2635 load_module: CAPI not installed. Sadly to say but those cards are not useable with capi. ISDN Cards for capi-driver are: -avm (fritz!, b1, c2, c4, some usb units) -eicon (diva-server-pci) -hypercope (dtmf not working, sound is crappy) Cards based on winbond are useable with i4l and IMHO mISDN. Donno if zaphfc would be useable right now or in near future!? lsmod | grep capi gives: capi 17472 0 capifs60242 capi kernelcapi46496 1 capi Anyone any suggestions of where to look? Anyone a working asterisk with ISDN on Debian? Groeten, Joost Kraaijeveld Askesis B.V. Molukkenstraat 14 6524NB Nijmegen tel: 024-3888063 / 06-51855277 fax: 024-3608416 e-mail: [EMAIL PROTECTED] web: www.askesis.nl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email fromBrekeke Announcing their RTP Proxy
I am not an OnDo user. Please do not spam me. - Original Message - From: SeshKanuri [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, September 25, 2004 4:47 AM Subject: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email fromBrekeke Announcing their RTP Proxy Dear Valued OnDO users, OnDO PBX v1.3 now supports 100 concurrent calls Brekeke is excited to announce our new OnDO PBX v1.3 with increased concurrent call capacity that is 4 times greater than the current release version OnDO PBX v1.2. -- How did Brekeke increase the capacity by so much? -- RTP relay OnDO PBX is well regarded among many users for its ease of use and administration. But we felt it needed to be spiced up to make our users and customers even happier. As you may know, Real-Time Transport Protocol (RTP) is commonly used for transmitting audio data for VoIP telephony. Until this version of OnDO PBX, all RTP packets were relayed through OnDO PBX. By minimizing the RTP packet traffic that go through OnDO PBX, we have succeeded in increasing the number of concurrent calls. You may ask how do we achieve this? By adopting DTMF-via-INFO method, we succeeded in decreasing the number of packets going through the server. All RTP packets are sent and received directly between SIP UAs, establishing peer to peer connections between them. Whenever an OnDO PBX user needs to send commands to forward calls or to put calls on hold, the information will be carried through SIP packets as INFO messages. For the phone or in situations where sending DTMF-via-INFO is not possible, OnDO PBX provides the option to turn RTP relay on or off from the OnDO PBX Admintool. -- Keep your Favorites with OnDO -- Brekeke offers NO restriction value to our customers Many VoIP products come with a series of strings attached, and the strings usually cost you more money. Brekeke strives hard to create products that won't restrict users' choice of phones, operating system platforms, or hardware equipment. As a result, we have a wide variety of users that use our products with their favorite list of products. With new OnDO PBX v1.3, we reserve the flexibility and openness of OnDO, yet enhance the possibility of OnDO products. OnDO PBX v1.3 is a super-sized version, so to speak. -- Questions about installing/using OnDO PBX? We know that it's a commitment on your part to learn about new software, so we'd like to help you! Brekeke Software offers a solution to technical support: an ONLINE FORUM THAT ALLOWS YOU TO DIRECTLY COMMUNICATE WITH OUR ENGINEERS. At Brekeke Software, we're always happy to help our users setup and configure their OnDO PBX systems for an optimal VoIP experience. Please post your technical questions in our support forum, and we'll get back to you as soon as we can. Brekeke Software is committed to offering you an unmatched level of support. http://www.brekeke.com/en/support/supportforum_en.html -- www.brekeke.com For sales information, please contact us at [EMAIL PROTECTED] Questions? Comments? Feedback? We'd love to hear from you! [EMAIL PROTECTED] Please do not reply to this email. You have received this email from Brekeke Software Inc because you registered to receive periodic news and updates that we believe may be of interest to you. TO UNSUBSCRIBE: Please send a blank email to [EMAIL PROTECTED] and you will be removed from our list. Copyright 2002-2004 Brekeke Software, Inc. All Rights Reserved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email fromBrekeke Announcing their RTP Proxy
Well maybe you should be a user. I offer much less than *, at only a much greater cost :-) I think this is a bit like advertising Windows XP on the Linux kernel mailing list :-) Regards, Steve Steve Totaro wrote: I am not an OnDo user. Please do not spam me. - Original Message - From: SeshKanuri [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, September 25, 2004 4:47 AM Subject: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email fromBrekeke Announcing their RTP Proxy Dear Valued OnDO users, OnDO PBX v1.3 now supports 100 concurrent calls Brekeke is excited to announce our new OnDO PBX v1.3 with increased concurrent call capacity that is 4 times greater than the current release version OnDO PBX v1.2. -- How did Brekeke increase the capacity by so much? -- RTP relay OnDO PBX is well regarded among many users for its ease of use and administration. But we felt it needed to be spiced up to make our users and customers even happier. As you may know, Real-Time Transport Protocol (RTP) is commonly used for transmitting audio data for VoIP telephony. Until this version of OnDO PBX, all RTP packets were relayed through OnDO PBX. By minimizing the RTP packet traffic that go through OnDO PBX, we have succeeded in increasing the number of concurrent calls. You may ask how do we achieve this? By adopting DTMF-via-INFO method, we succeeded in decreasing the number of packets going through the server. All RTP packets are sent and received directly between SIP UAs, establishing peer to peer connections between them. Whenever an OnDO PBX user needs to send commands to forward calls or to put calls on hold, the information will be carried through SIP packets as INFO messages. For the phone or in situations where sending DTMF-via-INFO is not possible, OnDO PBX provides the option to turn RTP relay on or off from the OnDO PBX Admintool. -- Keep your Favorites with OnDO -- Brekeke offers NO restriction value to our customers Many VoIP products come with a series of strings attached, and the strings usually cost you more money. Brekeke strives hard to create products that won't restrict users' choice of phones, operating system platforms, or hardware equipment. As a result, we have a wide variety of users that use our products with their favorite list of products. With new OnDO PBX v1.3, we reserve the flexibility and openness of OnDO, yet enhance the possibility of OnDO products. OnDO PBX v1.3 is a super-sized version, so to speak. -- Questions about installing/using OnDO PBX? We know that it's a commitment on your part to learn about new software, so we'd like to help you! Brekeke Software offers a solution to technical support: an ONLINE FORUM THAT ALLOWS YOU TO DIRECTLY COMMUNICATE WITH OUR ENGINEERS. At Brekeke Software, we're always happy to help our users setup and configure their OnDO PBX systems for an optimal VoIP experience. Please post your technical questions in our support forum, and we'll get back to you as soon as we can. Brekeke Software is committed to offering you an unmatched level of support. http://www.brekeke.com/en/support/supportforum_en.html -- www.brekeke.com For sales information, please contact us at [EMAIL PROTECTED] Questions? Comments? Feedback? We'd love to hear from you! [EMAIL PROTECTED] Please do not reply to this email. You have received this email from Brekeke Software Inc because you registered to receive periodic news and updates that we believe may be of interest to you. TO UNSUBSCRIBE: Please send a blank email to [EMAIL PROTECTED] and you will be removed from our list. Copyright 2002-2004 Brekeke Software, Inc. All Rights Reserved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
Michael Bielicki wrote: find someone to host it in India or serbia and you can safely ignore it :) On Sat, 25 Sep 2004 10:22:52 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 25 Sep 2004, Steve Underwood wrote: On Sat, 25 Sep 2004, Steve Underwood wrote: I am not a lawyer, nor even a US citizen. Talking to someone who is both may be a good idea. What is the relevance of being a US citizen? Copyright rules are largely global. There are two different major sets of copyright laws, depending on which treaty they were derived from. They are not always compatible. They differ in such points as whether you can transfer your copyright or merely assign the rights granted by it. There are a number of variants in this area. Some countries will not allow someone to put things into the public domain. Perhaps I should have been more specific. What is copyrightable is pretty much global. Unless otherwise granted by the copyright holder, by default the copyright of a derived work (in the copyright legan sense) is held by the owner of the original copyright and not the crator of the derived work. So no, the patches are owned by Intel as well. But the patches aren't a derived work. That is the value they have here. There are an independant adjunct work. According to most lawyers a patch _is_ a derived work in nearly all circumstances. E.g. a novel based on the characters from a novel by another author is a derived work. No. A patched copy is a derived work. A patch avoids containing enough of the original to count. If you are producing copies of just about anything you really need to speak to your lawyer to be safe. The excpetion possibly being open source stuff based soley on open source stuff. Anyway, this is getting too far off topic for this list. Mea culpa. Peter Keeping * legal is off topic? Weird notion. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to get Call Details Records
Title: Message HI, Can anyone please tell me 1) Where does asterisk store the call detail records? 2) What is thestructure of these call details records? 2)How to access the call detail records by any external application? Thanks in advance Regards, Mayank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email from Brekeke Announcing their RTP Proxy
Is it April 1st already, where did the year go Andy On 25/09/2004 at 01:47 SeshKanuri wrote: Dear Valued OnDO users, OnDO PBX v1.3 now supports 100 concurrent calls Brekeke is excited to announce our new OnDO PBX v1.3 with increased concurrent call capacity that is 4 times greater than the current release version OnDO PBX v1.2. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
On Sat, 25 Sep 2004, Steve Underwood wrote: But the patches aren't a derived work. That is the value they have here. There are an independant adjunct work. According to most lawyers a patch _is_ a derived work in nearly all circumstances. E.g. a novel based on the characters from a novel by another author is a derived work. No. A patched copy is a derived work. A patch avoids containing enough of the original to count. Well, you need to see your lawyer about that. What I said above is what the Usenix legal council told usduring a workshop. As an example, if I were to write a few more chapters to Gone With the Wind those would be a derived work and, in countries signatories to one of the two copyright treaties, the property of the original copyright holders. An explaination by someone more skilled with words than I am is at http://lists.ibiblio.org/pipermail/cc-licenses/2004-March/000528.html Had the patch been against the actual g729 libraries the case would have been clear. Now, the patch is against asterisk to make it interoperate with the g729 libarary and this may or may not be non-infringing. However, the distribution of the g729 libraries themselves are almost certainly infringing. There is also the possibility that the patch to asterisk may be ruled a contribuatory infringement. Just because it is a patch does not mean it is non-infringing. See a lawyer. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi install problem
Please can someone help me to install chan_capi on Mandrake 10. I get page after page of errors and can not seem to find detailed install instructions anywhere. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Only Accept Call After Pressing a Key '#' or '*'
I would like asterisk to dial an extension or external number but for the call to only be connected after the called party presses a key. Therefore been able to announce the call to the called party before answering. I have had this working on queued calls but want to incorporate this for standard dialled extensions. Our use for this would to be able to divert a call to a users mobile but only connect the call on the user answering the mobile phone and pressing a key after the announcment of the call. Im thinking this would get around the problem of asterisk considering the call answered when it actually goes to the mobile users voicemail. Therefore we could have a dial plan that calls several mobile phones but only connects if the user actualy acknowledges they have answered the call. Is this possible on a standard dial command? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Newbie configuration hell :-)
Sorry to post such a newb set of questions but I have been hammering
about trying to get Asterisk running on FC2 machine reading everything
available (I think that is what stuffed me, shouldn't have read it all
:-) ).
Config
FC2 running Asterisk 1.0.0, with the h323 compiled in and installed
correctly.
Amazingly enough I have everything compiled correctly and installed.
I am running a TDM400P, Port 1 FXS, Port 4 FXO.
I have my PSTN line plugged into 1 port and my Analogue phone plugged
into port 4 (I think that's right I get tone on the phone when I pick it
up and echo works).
/etc/zaptel.conf
fxols=1
fxsls=4
; Weird but I was told to have the fxols fxsls reverse to the actually
module
loadzone = au
defaultzone = au
/etc/zapata.conf
[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690 ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4
Extensions.conf
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
[outgoing]
exten = _1XX,1,Dial(H323/[EMAIL PROTECTED]) ; 1xx extension
to Salisbury
exten = _2XX,1,Dial(H323/[EMAIL PROTECTED]) ; 2xx extension
to Marcoola
exten = 610,1,Dial(H323/[EMAIL PROTECTED]) ; 610 to Jindalee
exten = 620,1,Dial(H323/[EMAIL PROTECTED]) ; 620 to Batteryhill
exten = _54XX,1,Dial(H323/[EMAIL PROTECTED]) ; 54 to Marcoola
exten = _0754XX,1,Dial(H323/[EMAIL PROTECTED]); 54 to
Marcoola
exten = _,1,Dial(Zap/g2/${EXTEN})
H323.conf
[general]
port = 1720
bindaddr = 192.168.69.1
tos=lowdelay
disallow=all
allow=g723.1
allow=gsm
--
I can pick up the phone and ring 099 and echo works but if I dial
anything else I just get a busy signal with no errors on asterisk
-c, what I need is for ANY incoming calls to make the analogue phone
ring.
Outgoing calls that fit the rules use h323, everything else should pick
up the PSTN line and dial.
I again apologise for the mess and newbness (did I just invent a word),
I just need a kick start and get the basic stuff working before I start
playing.
Also, anyone had asterisk talking to OKI Voip like BV1250 units
working?, if so can you drop me an email.
James
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Re: [Asterisk-Users] TDM400P Newbie configuration :-)
James Bean wrote:
Sorry to post such a newb set of questions but I have been hammering
about trying to get Asterisk running on FC2 machine reading everything
available (I think that is what stuffed me, shouldn't have read it all
:-) ).
Config
FC2 running Asterisk 1.0.0, with the h323 compiled in and installed
correctly.
Amazingly enough I have everything compiled correctly and installed.
I am running a TDM400P, Port 1 FXS, Port 4 FXO.
I have my PSTN line plugged into 1 port and my Analogue phone plugged
into port 4 (I think that's right I get tone on the phone when I pick it
up and echo works).
/etc/zaptel.conf
fxols=1
fxsls=4
; Weird but I was told to have the fxols fxsls reverse to the actually
module
loadzone = au
defaultzone = au
/etc/zapata.conf
[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690 ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
Here you have a context of pstn, which I assume is your incoming dialtone.
channel=4
Extensions.conf
But where is the pstn context in Extensions to match the above incoming
dialtone?
Mayb you want something like this:
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM})) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above
exten = s,3,VoiceMail(u100) ;Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
[outgoing]
exten = _1XX,1,Dial(H323/[EMAIL PROTECTED]) ; 1xx extension
to Salisbury
exten = _2XX,1,Dial(H323/[EMAIL PROTECTED]) ; 2xx extension
to Marcoola
exten = 610,1,Dial(H323/[EMAIL PROTECTED]) ; 610 to Jindalee
exten = 620,1,Dial(H323/[EMAIL PROTECTED]) ; 620 to Batteryhill
exten = _54XX,1,Dial(H323/[EMAIL PROTECTED]) ; 54 to Marcoola
exten = _0754XX,1,Dial(H323/[EMAIL PROTECTED]); 54 to
Marcoola
exten = _,1,Dial(Zap/g2/${EXTEN})
H323.conf
[general]
port = 1720
bindaddr = 192.168.69.1
tos=lowdelay
disallow=all
allow=g723.1
allow=gsm
--
I can pick up the phone and ring 099 and echo works but if I dial
anything else I just get a busy signal with no errors on asterisk
-c, what I need is for ANY incoming calls to make the analogue phone
ring.
See comment above.
Outgoing calls that fit the rules use h323, everything else should pick
up the PSTN line and dial.
I again apologise for the mess and newbness (did I just invent a word),
I just need a kick start and get the basic stuff working before I start
playing.
Also, anyone had asterisk talking to OKI Voip like BV1250 units
working?, if so can you drop me an email.
No idea on that.
--
respectfully, Joseph
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[Asterisk-Users] RE: CTI development
Or what is it that you meant in particular? I'l bet he means 3rd party call control like in a traditional CTI deployment ala Cisco ICM, Genesys or an oldie-but-goodie, IBM CallPath DirectTalk. (Net-net version) Basically, a scratch-pad type area of ~2K that gets created/destroyed with every call and _follows_ the call for its life in the system. Olus the ability of a 3rd party computer application aka softphone to control the telephone appliation - this part we've got but still needs some modification for true CTI. (Example) So the caller gets to the IVR. The IVR pushes data relevant to the current call onto the scratch pad using a unique call event ID then xfers the call to the call centre Q. The call gets allocated to an agent in the Q. Their desktop application gets an alerting message which is basically a ring event alerting them that they are about to get the next event including the internal ID of the event. (In traditional environments this happens _slightly_ before the phone rings. The application then reads the scratch pad data associated with the call event ID so the desktop can have full context of what has gone before in the call. The desktop application then does whatever it needs to do in the customer environment - this is custom development - the CTI vendor offers an SDK for interface to their softphone product. The desktop application needs the ability to also write/update to the scratch pad as there may be a need to xfer the call to another agent or back to the IVR which should be able to read the updated data. I may not have the skill to code all of the application, but I'm a call centre solution architect. If anyone would like to bring this functionality to Asterisk I would be excited to offer industry advice. There are lots of gotchas in the CTI world that are completely _not_ related to programming skill. The wrong implementation simply won't have a market. dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Sarge, ISDN, CAPI and Asterisk blues
Hi Joost, the W6692 based cards do NOT have capi drivers. At least not with isdn4linux, maybe it would work with the mISDN drivers. I have a W6692 card laying around on my desk (thanks voidptr :) ), a zaptel driver for that chipset is planned, but of course other things are more important. ;) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-09-25 um 10.12 schrieb Joost Kraaijeveld: Hi all, I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest testing versions). If I start asterisk I get: chan_capi.c:2635 load_module: CAPI not installed. lsmod | grep capi gives: capi 17472 0 capifs60242 capi kernelcapi46496 1 capi Anyone any suggestions of where to look? Anyone a working asterisk with ISDN on Debian? Groeten, Joost Kraaijeveld Askesis B.V. Molukkenstraat 14 6524NB Nijmegen tel: 024-3888063 / 06-51855277 fax: 024-3608416 e-mail: [EMAIL PROTECTED] web: www.askesis.nl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
On Saturday 25 September 2004 06:03, Peter Svensson wrote: As an example, if I were to write a few more chapters to Gone With the Wind those would be a derived work and, in countries signatories to one of the two copyright treaties, the property of the original copyright holders. IANAL, but those chapters would be yours. Adding them to Gone With the Wind and distributing the resultant new book would be considered distributing a derived work and fall into the gray area. It's the same as fanfic; the characters and whatnot are owned by the original writer(s) but your actual work is yours. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Groups
Hi! The first hurdle you must take is finding out what busy exactly means for your SIP phones - do you allow only 1 call appearance, or 2, or ... see the dialplan commands SetGroup, GetGroupCount etc. for this. Note: Before this feature was added to Asterisk people used outgoinglimit= and incominglimit= in sip.conf. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup I've looked around trying to find a solution to this problem but I haven't found anything that works quite the way I want it to. I know you can use Dial(SIP/0SIP/1SIP/2,20,Ttr) to dial all three extension at the same time but this won't work for me. I also know that I could set up a dial plan to go from one extension to the next but I only want the phone to ring a max of 4 to 6 times. Also, I imagine I could use call queues but this is supposed to be a Reception phone and that doesn't seem to fit here. Why not simply use the queue if that solves your problem? You'll need to transalte ring 4-6 times into a value in seconds, but that you can manage I assume... :-) Finally: If you set up the dial plan to go from extension to extension you'll get exactly what you want. Asterisk knows immediately if a phone is busy or not (limitations see above), so you are not wasting time (or x rings). If you like you can add a 'non-busy call attempt' counter using the SetVar dialplan command coupled with GotoIf() to prevent trying yet another extension... Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Put Asterisk 1.0 mirrors into the Wiki
Hi folks, I'd like to encourage all of those friendly mirror maintainers to include their link here in the appropriate place: http://www.voip-info.org/wiki-Asterisk-mirrors Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
I was really looking forward to Asterisk 1.0 et al, but it is a major disappointment. I have never experienced any Asterisk release that was interacting with Digium hardware so unreliably. Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as the call is being picked up at the other end. I have tried various X100P (original Digium) cards, various phone lines and just about every possible combination of parameters in zaptel.conf (from every feature off to most features on) but without any success. The same hardware was working fine with CVS from about a month ago. The problem of false hangups really needs to be fixed. A false hangup is NEVER EVER acceptable in an office environment. On the other hand, a call that doesn't hangup even if the remote party has already hung up is ALWAYS acceptable. Therefore, if the software is not capable of detecting hangups properly, then why not provide a setting to disable any and all hangup decisions now made by software and let the human user decide instead. There should be a setting hangup=local-only that would have the effect that no channel will ever be hung up unless the (non-Zap) local party has hungup. As things stand now, we won't be able to deploy this 1.0 release if Zaptel is required. What a pity. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Sarge, ISDN, CAPI and Asterisk blues
On Sat, Sep 25, 2004 at 11:18:23AM +0200, Thomas Niesel wrote: Donno if zaphfc would be useable right now or in near future!? Worked fine for me here with $20 card until entire Alcatel pbx locked up and they blamed our line.. arkadi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi module
Title: Nachricht Sorry, I cant help, but I do have the exact same problem with compiling chan_capi module under RH 9.0. Anybody any Idea? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 and Asterisk intellectual property issues
-- snip -- Had the patch been against the actual g729 libraries the case would have been clear. Now, the patch is against asterisk to make it interoperate with the g729 libarary and this may or may not be non-infringing. However, the distribution of the g729 libraries themselves are almost certainly infringing. There is also the possibility that the patch to asterisk may be ruled a contribuatory infringement. -- snip -- The patch is not against Asterisk - it is against Intel's sample code. No parts of Asterisk are modified in order to run this code. Nor am I requesting that Asterisk be modified in any way to support this. The code produced by running the build script is a shared library that can be added to Asterisk. The shared library could be used independently of Asterisk, and Asterisk can still be used without the shared library. It is completely optional whether people choose to integrate this code with Asterisk. However, I understand that it probably can't be added to the main distribution and I am happy to continue making it available in source form as an add-on module for those who would like to evaluate it. I certainly never expected that it would be adopted as an official inclusion in Asterisk, and I certainly won't take offence if it isn't. The relevent terms from Intel's license are below. (B) says that I have the right to modify the source code and (C) says that I can combine portions of the sample source into a product and then distribute the resulting application. B. Subject to all of the terms and conditions of this Agreement, Intel grants to you a non-exclusive, non-assignable copyright license to modify the Materials, or any portions thereof, that are (i) provided in source code form or, (ii) are defined as Redistributables and are provided in text form. C. Subject to all of the terms and conditions of this Agreement, Intel grants to you a non-exclusive, non-assignable copyright license to distribute (except under an Evaluation License as specified below) the Redistributables and Sample Source, or any portions thereof, as part of the product or application you developed using the Materials. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
On Sat, 25 Sep 2004, Andrew Kohlsmith wrote: IANAL, but those chapters would be yours. Adding them to Gone With the Wind and distributing the resultant new book would be considered distributing a derived work and fall into the gray area. It's the same as fanfic; the characters and whatnot are owned by the original writer(s) but your actual work is yours. Most fanfics are probably owned by the original author/distribution company. However, they are not stupid enough to try to enforce their rights since there would probably be a public backlash. The fanfics probably generates increased revenue for them as well. Think cheap advertising. See http://www.templetons.com/brad/copymyths.html, point 6. Creating a derived work is the a right only available to the copyright owner and any licensees of that right. Derived work is a very broad term. The ownership of derived works I have a harder time to find references to. The closest thing is that apparently there is no copyrights available to the author of a derived work which is unlawfully made from a copyrighted original work. And now I will shut up lest someone takes my word for anything instead of seeking legal counsel. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) Cheers, Arik -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBVWUr//PXyz2NiW8RAuhdAJ0RPC1uwm82HJsz8/t1WMwZLi1D/QCeNu++ IrelQXpR1JpkP18t6a+K4tk= =vtbn -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
Am Sa, 2004-09-25 um 14.31 schrieb Arik Funke: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) Cheers, Arik Hi, 22 MB zipped for an *, postfix, router, traffic shaper, sshd. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
On Saturday 25 September 2004 08:31, Arik Funke wrote: I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) I will say this to every single person who comes in here asking what the BARE MINIMUM is or HOW MUCH can Asterisk handle... You do *not* know enough about the system to even attempt to build these kinds of systems! In order to properly provision Asterisk, you need experience with it. You need to know how it operates normally, and how it operates when it's struggling. Build a normal Asterisk box first. Play with it. Get to know it. THEN start optimizing. PLEASE -- will people stop trying to optimize their Asterisk system until they have Clue One about how it operates and what its requirements are? I am asking that you do this for your own good; I want you to have a successful Asterisk install and blindly telling you is NOT going to help you achieve that in any way shape or form. PLEASE -- UNDERSTAND THE SYSTEM, THEN OPTIMIZE. NOT THE OTHER WAY AROUND. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
On Saturday 25 September 2004 08:03, Benjamin on Asterisk Mailing Lists wrote: Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as the call is being picked up at the other end. Disable callprogress and/or busydetect. I have tried various X100P (original Digium) cards, various phone lines and just about every possible combination of parameters in zaptel.conf (from every feature off to most features on) but without any success. Where are you located? Is your phone company perhaps giving you call completion supervision? Your post is quite short on details. The problem of false hangups really needs to be fixed. A false hangup is NEVER EVER acceptable in an office environment. On the other hand, a call that doesn't hangup even if the remote party has already hung up is ALWAYS acceptable. Therefore, if the software is not capable of I disagree wholeheartedly; call startup and termination must both be reliable. Having 5 minutes of busy tones on VM is not acceptable. So let's get to the root of your problem. user decide instead. There should be a setting hangup=local-only that would have the effect that no channel will ever be hung up unless the (non-Zap) local party has hungup. You must be new here. As things stand now, we won't be able to deploy this 1.0 release if Zaptel is required. What a pity. With that attitude, we won't even miss you. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM phones, bluetooth and general happiness
We've been using the CellSocket on asterisks in our lab and it works well. They only problem we found was DTMF performance from the local cell phone to asterisk has varied depending on carrier and phone model. /ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William Suffill Sent: Friday, September 24, 2004 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM phones, bluetooth and general happiness Interesting. I think either the phonelabs adapter or cellsocket might be an interesting idea. We are moving to a biz mobile package I use iax2 term to fwd to a nextel since it's free inbound but having a cell on the asterisk box is probably a better fit. Besides on a biz plan w/ tmobile and others you can add a line for $10 on the pooled mins plans. Very interesting idea ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp.so: _use_ast_pthread_create_instead_
On Sat, Sep 25, 2004 at 06:48:02AM +0200, Goran Dj. arranged a set of bits into the following: I tried to install chan_sccp (make; make install) but after that when asterisk starting: [chan_sccp.so]Sep 25 06:34:28 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: __use_ast_pthread_create_instead__ Sep 25 06:34:28 WARNING[16384]: loader.c:423 load_modules: Loading module chan_sccp.so failed! I tried to replace pthread_create() with ast_pthread_create() in chan_sccp.c, but same error... Help? Use CVS chan_sccp, it has the fix for this (and other changes). Anon CVS access is easy using the information on the sccp site. http://chan-sccp.sf.net/ Also that seems to indicate that you were compiling chan_sccp against a different version of asterisk then you are running (this may not be so, but please check). Thank, Julien (chan_sccp developer) pgpsqgQJv8rBm.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ilbc problem
Hello, I'm going to use * as SIP-H.323 proxy (codecs doesn't matter - only pass through). I compile * (v1.0.0) without any problems as far as H.323 stack (pwlib, etc). But when I'm trying execute asterisk -vvv I'm getting error message: [codec_ilbc.so]Sep 25 15:15:43 WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/codec_ilbc.so: undefined symbol: sqrt Sep 25 15:15:43 WARNING[16384]: loader.c:429 load_modules: Loading module codec_ilbc.so failed! Is it possible to compile Asterisk wo. any codecs, or what's the easiest way to solve this problem? (We are using AudioCodes hardware to terminate VoIP into PSTN). Distr. - Debian Woody 3.0, libc6 2.3.2, kernel 2.4.26 -- Marcin Kwiatkowski http://www.telebonus.pl/ Telebonus Sp. z o.o. 43-300 Bielsko-Biaa ul. Legionw 30 pho.: +48 (33) 819 49 66 mob.: +48 605 923 944 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with dialing out with TDM400P
Scenario,
I got some very good help earlier from Joseph getting me up and started
but I have a couple of small problems still.
Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4
Analog dialout line and Analog handset plugged in.
Problems:
1.
Incoming calls work and the phone rings and can be answered no problems,
(although I wouldn't mind being able to adjust the ring but that's not
important), I can't ring out, I just get a busy signal and nothing comes
up on the console. I am pretty sure its just a simple line missing from
extensions.conf.
2.
I am based in australia and when I have an incoming call with callerid
turned on then I get the following error on console.
-- Zap/1-1 is ringing
Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event:
Didn't finish Caller-ID spill. Cancelling.
---
/etc/zaptel.conf
fxols=1
fxsls=4
loadzone=au
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above
#exten = s,3,VoiceMail(u100);Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test
/etc/asterisk/zapata.conf
[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690 ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4
---
Any help would be very much appreciated.
James
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RE: [Asterisk-Users] agents and queues
-Original Message- From: Marco Nicolayevsky [mailto:[EMAIL PROTECTED] Sent: Friday, September 24, 2004 11:45 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] agents and queues How can i determine if there are any agents signed-in, and if not, take them straight to voice mail with a message like Sorry, we are unable to take you call now, please leave a message...?? We have run into the same sort of problems. We created an addition to the monastery project which allows our call center manager to see who is logged in via a web browser. In that same perl script which loops indefinately every 5 sec we check to see if there is anyone logged in. Then we create a variable (AGENTSLOGGEDIN) which is either 0 or 1. Then we check the status of that variable from the dialplan to see if we should place calls in the queue. Seems to work pretty well for us. Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon Developers Conference Recordings
I've have the main Astricon dev conference from 12PM to the end recorded and posted at http://snipurl.com/astricon . Due to overloaded hotel uplink (T1) there are some spots with no audio where the uplink droped out for a few minutes. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to get Call Details Records
Title: Message see: http://www.voip-info.org/wiki-Asterisk+billing - Original Message - From: Mayank Mishra To: [EMAIL PROTECTED] Sent: Saturday, September 25, 2004 2:10 PM Subject: [Asterisk-Users] How to get Call Details Records HI, Can anyone please tell me 1) Where does asterisk store the call detail records? 2) What is thestructure of these call details records? 2)How to access the call detail records by any external application? Thanks in advance Regards, Mayank ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with dialing out with TDM400P
Incoming calls work and the phone rings and can be answered
no problems, (although I wouldn't mind being able to adjust
the ring but that's not important), I can't ring out, I just
get a busy signal and nothing comes up on the console. I am
pretty sure its just a simple line missing from extensions.conf.
In your [internal] context try something like..
exten = _0.,1,Answer
exten = _0.,2,Dial(Zap/g1/${EXTEN:1})
exten = _0.,3,Hangup
This way Asterisk will send all the digits dialled after the 0 to the
zaptel card and you should be dialing out. You may not need the
answer/hangup lines for your setup.
2.
I am based in australia and when I have an incoming call with
callerid turned on then I get the following error on console.
-- Zap/1-1 is ringing
Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event:
Didn't finish Caller-ID spill. Cancelling.
I'm not sure if this is related with inbound CallerID on an FXO, but to
get Caller ID working on an FXS port I had to make this change to the
chan_zap.c file and recompile:-
http://lists.digium.com/pipermail/asterisk-users/2004-August/057349.html
In /usr/src/asterisk/channels/chan_zap.c
#define DEFAULT_CIDRINGS 2
The default is 1.. Seems we need this set to 2 in Australia, I dare say
making this change might get the inbound caller ID working for you also.
Hope this helps,
Chris Lee
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[Asterisk-Users] getting variable using agi
Hi I'm try to get any variable (i.e.:CALLERID) on my agi script in perl. Using the function get_variable(), the value is empty... I read that the function don't work properly... Please, ignore my terrible english (i'm from 'sao jose dos campos', brazil). Thanks, Ricardo Maia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi install problem
On Sat, 2004-09-25 at 11:43 +0100, Nicolas Whitham wrote: Please can someone help me to install chan_capi on Mandrake 10. I get page after page of errors and can not seem to find detailed install instructions anywhere. So you phone the AA or RAC and say my car's stopped and nothing else, where do you expect to be put on the priority list. To allow _anyone_ to help you you'll have give more information. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Developers Conference Recordings
On Sat, 25 Sep 2004 07:00:46 -0700, Brian [EMAIL PROTECTED] wrote: we have a mirror for that at: http://astricon.asterisk.pl/2004-09-recordings/index.php -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
On Sat, 25 Sep 2004 21:03:31 +0900, Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: The problem of false hangups really needs to be fixed. A false hangup is NEVER EVER acceptable in an office environment. On the other hand, a call that doesn't hangup even if the remote party has already hung up is ALWAYS acceptable. Therefore, if the software is not capable of detecting hangups properly, then why not provide a setting to disable any and all hangup decisions now made by software and let the human user decide instead. There should be a setting hangup=local-only that would have the effect that no channel will ever be hung up unless the (non-Zap) local party has hungup. I think that letting the non-Zap handle the hangup is a very good idea. If the non-Zap originated the call, than it has the right to terminate it. If it didn't, it will eventually perceive that the caller hung up, and won't spend the day on a dead phone... It is better than the call being disconnected in the middle of an important discussion, and it may create the impression that the other person slammed the phone on you if you were arguing or something like that... The only problem is incoming calls to IVR, VM, and such. It should be allowed to specify if busy detect is enabled only for zap-originated, zap-terminated, or both kinds of calls. That's my opinion... :) Marconi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ilbc problem
On Sat, 25 Sep 2004 15:18:27 +0200, Marcin Kwiatkowski [EMAIL PROTECTED] wrote: [codec_ilbc.so]Sep 25 15:15:43 WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/codec_ilbc.so: undefined symbol: sqrt Sep 25 15:15:43 WARNING[16384]: loader.c:429 load_modules: Loading module codec_ilbc.so failed! Is it possible to compile Asterisk wo. any codecs, or what's the easiest way to solve this problem? (We are using AudioCodes hardware to terminate VoIP into PSTN). The easiest way: in modules.conf, add: noload = codec_ilbc.so (check for misspelling) or delete codec_ilbc.so from the modules dir... Caveman-style :) Marconi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs Problem?
Hello, I have a following setup: IP phone (Cisco/Skinny) - * - NAT -- NAT - * - PSTN Everything is perfect when i'm using it from right to left. From left to right however, there is no voice, although the calls are being placed. I played around with codeces but no change. Does anybody know, what I possibly am doing wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco PIX and Asterisk
I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some photos from Astricon 2004
el Flynn wrote: Lenny Tropiano / asterisk.org Mailing list wrote: These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny Anyone knows if those Snom Keypad 220s are available, and where I might be able to get my hands on a few? I was talking to NETXUSA at the show, and they have them in stock. They also had them set up and working (though they hadn't tried the BLF (Busy Lamp Field) aspect of them when I checked. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
On 25/09/2004 at 14:31 Arik Funke wrote: Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) Cheers, Arik You could start buy downloading my .iso (29mb bootable ) and use that as a basisis for your system. I've already modified it for a CF card based system. Essentially it depends what sort of interface to the pstn you want. E1/T1 and analog should work fine with my cd - but I've not built it for use with CAPI or the QuadBRI cards... you can grab it at http://www.automated.it/asterisk/ It's not v1 of * but I am trying to find the time to update to a newer CVS version, however I will only do that once I'm happy running that particular version myself... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
On Sat, 25 Sep 2004 08:45:56 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: Disable callprogress and/or busydetect. I wouldn't have posted without having tried that beforehand. The problem persists with both busydetect and callprogress disabled. Where are you located? In Japan. Lines are provided by NTT. The driver (wcfxo.o) has been built with #define JAPAN uncommented. Before the present 1.0 release, this has usually reduced false hangups. Is your phone company perhaps giving you call completion supervision? No. I disagree wholeheartedly; call startup and termination must both be reliable. Having 5 minutes of busy tones on VM is not acceptable. Fair enough. But then again, why not have an option that disables hangup detection until a call actually goes to voicemail and leave it disabled if it doesn't?! Anyway, so far false hangups with Zaptel on Japanese phone lines have been mostly a sporadic problem, but with this release Asterisk hangs up *every time*. As I have said in my earlier post, the same hardware did not have this problem with CVS from about a month ago (August, 19 or 20). rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P FXO and Primus TalkBroadBand
Ryan Courtnage wrote: Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus -- DLink ATA FXS -- TDM400P FXO -- Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS -- TDM FXO link 'goes dead'. Has anyone had any measure of success doing this? Primus' service is becoming very popular in Canada, and some customers are wanting to do this. Not with Primus/Dlink, but I am having the same issue with by Vonage/Motorola. I have not really looked into it yet, though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with dialing out with TDM400P
I don't see anything posted here in extensions.conf to allow dialing out on
group 2.
You need something like this:
[outgoing]
exten = _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9X.,2,Congestion()
And add the context outgoing to those extensions that you allow to dial out
to the PSTN.
Lyle
- Original Message -
From: James Bean [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 8:28 AM
Subject: [Asterisk-Users] Help with dialing out with TDM400P
Scenario,
I got some very good help earlier from Joseph getting me up and started
but I have a couple of small problems still.
Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4
Analog dialout line and Analog handset plugged in.
Problems:
1.
Incoming calls work and the phone rings and can be answered no problems,
(although I wouldn't mind being able to adjust the ring but that's not
important), I can't ring out, I just get a busy signal and nothing comes
up on the console. I am pretty sure its just a simple line missing from
extensions.conf.
2.
I am based in australia and when I have an incoming call with callerid
turned on then I get the following error on console.
-- Zap/1-1 is ringing
Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event:
Didn't finish Caller-ID spill. Cancelling.
---
/etc/zaptel.conf
fxols=1
fxsls=4
loadzone=au
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above
#exten = s,3,VoiceMail(u100);Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test
/etc/asterisk/zapata.conf
[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690 ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4
---
Any help would be very much appreciated.
James
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RE: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andy Powell Sent: 25 September 2004 16:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway On 25/09/2004 at 14:31 Arik Funke wrote: Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) Cheers, Arik You could start buy downloading my .iso (29mb bootable ) and use that as a basisis for your system. I've already modified it for a CF card based system. Essentially it depends what sort of interface to the pstn you want. E1/T1 and analog should work fine with my cd - but I've not built it for use with CAPI or the QuadBRI cards... you can grab it at http://www.automated.it/asterisk/ It's not v1 of * but I am trying to find the time to update to a newer CVS version, however I will only do that once I'm happy running that particular version myself... HTH Andy Andy, I would be interested in a CF version too. Please, keep us posted on any progress. Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Whoa.... I'm owned but found ??
I get this message at CLI. what does it mean? - shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
On Sat, 25 Sep 2004 11:41:12 -0300, Marconi Rivello [EMAIL PROTECTED] wrote: It is better than the call being disconnected in the middle of an important discussion, and it may create the impression that the other person slammed the phone on you if you were arguing or something like that... More importantly, you will not get any customer to sign acceptance for an Asterisk system if they have false hangups. This is one of the things that customers will simply not accept, and rightly so. It is also one of those things that will go around very quickly and have the potential to damage Asterisk's reputation. Sure, you may say that if one want's to be assured there are no false hangups one should go for PRI. However, in this market over here, this is not an option, at least not yet. Then again, customers will simply tell you that they didn't have any false hangups on analog lines with directly connected analog telephone sets. They will say, if those ordinary analog phones don't hangup, then a PBX shouldn't have a problem either. It's the customers who make the rules, not us. The only problem is incoming calls to IVR, VM, and such. IVR doesn't have to be a problem, because you can program time-outs into your IVR menus. For voicemail, indeed, you'd want some software driven hangup detection, but since when do we subscribe to the all or nothing philosophy? Why not enable hangup detection selectively, ie only upon sending a call to voicemail? At least as an option! It should be allowed to specify if busy detect is enabled only for zap-originated, zap-terminated, or both kinds of calls. That would be better than all or nothing, but since we are talking about an option here, where is the harm to *also* provide a setting that disables detection outright, then provide selective means to enable it in the dialplan depending on context and/or call flow. You could then enable far-end detection for voicemail, local detection only for person-to-person calls, and for IVR calls as you see fit, depending on whether you have time-outs or not. And those who are in the lucky position not to have any false hangups, they would simply leave the setting on default and everything stays as it is now. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco PIX and Asterisk
I have a customer that wants to try the exact same thing next month. Unfortunately I dont have any advice for you at this time. However, if the PIX doesnt end up working for you I can tell you that Ive had excellent success with the INGATE product line. (Both Firewall and Firewall Traversal products) Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: Saturday, September 25, 2004 8:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco PIX and Asterisk I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
On Saturday 25 September 2004 11:28, Benjamin on Asterisk Mailing Lists wrote: Disable callprogress and/or busydetect. I wouldn't have posted without having tried that beforehand. Fair enough, I saw that you'd written tried every option but a lot of people don't actually mean that. :-) Where are you located? In Japan. Lines are provided by NTT. The driver (wcfxo.o) has been built with #define JAPAN uncommented. Before the present 1.0 release, this has usually reduced false hangups. Hmm okay so it is a known bug then; have you done any hunting around on the bugtracker or bothered a bug marshall? That would be my next step, and possibly where I'd expect to sit and hang for a while unless you could get the attention of someone with the skills to really dig in and fix it. Fair enough. But then again, why not have an option that disables hangup detection until a call actually goes to voicemail and leave it disabled if it doesn't?! Becuase it's a workaround and doesn't actually address the problem? In your case it might be a valid solution though; I wonder how hard it'd be to actually hack in? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco PIX and Asterisk
Are any packets at all from the incoming call setup getting though the PIX? In general, static NAT (plus access list), is required to enablean endpont with a global IP address to establish a connection to an endpoint behind the PIX with a private IP address. Are you using static NAT and what version of PIX OS are you running? John Chad Brown [EMAIL PROTECTED] wrote: I have a customer that wants to try the exact same thing next month. Unfortunately I don’t have any advice for you at this time. However, if the PIX doesn’t end up working for you I can tell you that I’ve had excellent success with the INGATE product line. (Both Firewall and Firewall Traversal products) Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig WaddingtonSent: Saturday, September 25, 2004 8:17 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Cisco PIX and Asterisk I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) à Internet à PIX (NAT) à Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta.___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free G.729 ready for download
There's another legal side to all of this which we need to evaluate carefully. Putting the list and Digium, at risk, by being in a position of having it used to break the law. Starting a few years ago ISPs became liable for harboring lawbreaking customers, and ended up answering to the court. If a court can be convinced that a particular list is used to spread illegal copies of let's say G729, then it's possible it could be held liable. The only thing I see missing from those types of court cases at this point, is Digium have probably not received a letter saying their customers are using their resources to violate someones copyright/patent - with a cease and desist letter. So the question is - do we really want to take that chance? Lawers do what they do as that is their livelyhood. If we get someones attention once, it will be that much closer to a second time. The law breaking would be trafficing in illegal copies of G729 with the intent of breaking the law. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free G.729 ready for download
steve szmidt wrote: The law breaking would be trafficing in illegal copies of G729 with the intent of breaking the law. Clearly, there is ample evidence in the list archives that the members of the list strove valiantly, in the face of greatly confusing and generally burdensome IP laws, to make sure they were in compliance with patent law. I don't think this circumstance justifies the somewhat hysterical reaction you gave. I know This is The Land of the Patriot Act, but I am doubtful we'd find a law anywhere on the books captioned as you present it above. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 Zaptel 1.0 -- False Hangup Disaster
On Sat, 25 Sep 2004 12:41:03 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: Fair enough, I saw that you'd written tried every option but a lot of people don't actually mean that. :-) :-) Lines are provided by NTT. The driver (wcfxo.o) has been built with #define JAPAN uncommented. Before the present 1.0 release, this has usually reduced false hangups. Hmm okay so it is a known bug then; Not sure whether it would have been considered a bug before. If you look in the wcfxo code, the JAPAN define seems to simply set some values differently, ie offhook-debounce or whatever it's called. have you done any hunting around on the bugtracker or bothered a bug marshall? I have put it on the bugtracker, but Mark has declared it resolved for technicality reasons I don't fully understand, which is not to say that I am criticising it. I was lucky enough to catch Mark on the chat and we talked about how to approach this. He told me that 1.0 was a snapshot of last Thursday's or Friday's CVS and that I should find out exactly when between 20 Aug and 1.0 this broke. So I will have a bit of testing to do over the next days. possibly where I'd expect to sit and hang for a while I had hoped to use 1.0 for an upcoming customer deployment because I thought it was a decendent of RC2 and there was a feature freeze, but now that I know it's just a very recent CVS I am not so sure I can dive in head over heels, so I will use the CVS of May 1st which has proven to be rather stable and with no or few surprises. But I'll dedicate a machine or two to 1.0 testing. Fair enough. But then again, why not have an option that disables hangup detection until a call actually goes to voicemail and leave it disabled if it doesn't?! Becuase it's a workaround and doesn't actually address the problem? In your case it might be a valid solution though; I wonder how hard it'd be to actually hack in? Well, that's what I was wondering about. Workaround or not, if it makes a huge difference for customers who would otherwise shun Asterisk, then it shouldn't be too much of a religious concern. Besides, I was suggesting this as an *option* that would by default be disabled, so it wouldn't make any difference for those who have no false hangups. Anyway, we'll first have to find the culprit. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Non-PRI T1 configuration
I'm trying to hook up a non-PRI fractional T1 using a T400P port. The
Telco says that it is provisioned as AMI with SF (not ESF) and that they are
signalling by sending down a straight DS1 (I'm not sure what exactly that
means). They are also sending DNIS over these channels. I currently run it
through a channel bank for my IVR application and it works fine but I am now
trying to convert to *.
This leaves me with three questions. First, * does not have an option for
SF framing. If I use ESF, should that work or is there another way?
Second, how do I configure the channel signalling in both zaptel.conf and
zapata.conf?
Third, how can I capture the DNIS in this situation or will it
automatically be available in the ${EXTEN} variable and also passed to AGI
scripts?
I would appreciate any help.
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[Asterisk-Users] Cisco 7960 and Asterisk...not working...
Hi! I have Asterisk up and running and have KPhone, IAXtel and FWD connected to it and working. I have a Cisco 7960 with version 3 (App. Load ID POS3-2-00) software. I have configured the 7960 correctly, I think; I have set everything - name, shortname, auth.name and display name set to 200. I have set the password to 200. I've set the proxy address/port to 192.168.1.117/5060. I can't seem to get the phone to connect to Asterisk, though Kphone works fine. Does anyone have an idea of what I am doing wrong? TIA, Chuck Wegrzyn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email from Brekeke Announcing their RTP Proxy
On Sat, 25 Sep 2004 01:47:38 -0700, SeshKanuri [EMAIL PROTECTED] wrote: Dear Valued OnDO users, ? [snip] For sales information, please contact us at [EMAIL PROTECTED] You have received this email from Brekeke Software Inc because you registered to receive periodic news and updates that we believe may be of interest to you. TO UNSUBSCRIBE: Please send a blank email to [EMAIL PROTECTED] and you will be removed from our list. This has SPAM stamped all over it and the claimed opt-in is outright untrue. SeshKanuri, you better be more careful because this could well lead to your ISP canceling your internet access, apart from making a fool of yourself before everybody on this list. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue and Agent functionality
I've seen alot of posts lately on Queue and Agent functionality, and alot of hacks to make them do different things that most call center managers want. In the sake of doing this one time, I'd like to develop a single list of request so we can consolidate a feature request for the Queue/Agent system. Here are the ones that I run into the most: 1. Queue should know the status of agents assigned to a queue and act accordingly. Here are a couple examples of the problem. A queue has no agents logged in and handling the queue, a call comes in for the queue, the call remains in the queue until either an agent logs in, or the queue reaches it's timeout. What it should do is immediately time out setting priority +101. Normal timeout (caller in busy queue with agents active) should exit with priority set +1. A Queue has active agents in a prioritized fashion. Agent 1 is priority 1, 2 is 2, 3 is 3, and 4 is 4. Agent 1 needs to make an outbound call as does agent 2. Both are now 'busy'. The Queue still attempts to call agent 1, gets 'busy' back from the sip device (i've only tried this with sip), and then the system appears to wait for something like 7-8 seconds before trying the next agent in line. 2. The queue system should allow a set of messages to be played at specific times. For example, a message that is played upon entry into the queue and no other time, the current set of messages played every frequency=XX, a message played to the caller when the call is accepted by an agent (eg transfering), finally, a set of messages played to the user based upon a predefined period int he config file.. see example below message1-time=time in seconds message1-frequency=never|once|always message1=message1-file-loc message2-time=time in seconds message2-frequency=never|once|always message2=message2-file-loc Where a message messageX-file-loc is played never|once|always every time in seconds. if time is set to 0, or freqency is set to never, the message is not played. If time is set to 0, and frequency is set to once, message is played at messagex-time, and never again. if time is set to 0 and frequency is set to always, message is played every messagex-time in seconds. 3. Agent timeout (logs the agent off if they do not respond to a ring in a defined about of time) does not track across calls. For example, if an agent steps away and forgets to log out, then thier phone will ring based upon whatever call strategy is used. If the agent timeout is set higher than the time the queue polls a set of agents they will never be logged out. The timer needs to increment per agent across multiple polls. So if my queue poll timer is 20 secons, but the agent timeout is set to 60 seconds, the preferred function would be to log the agent out of the queue if they completely miss three poll events. 4. If a caller empties a handled queue (active agents) with no callers, the caller will still hear messages (you are first in queue, etc.). This should not occur. Someone posted a 2-line patch on -dev list recently to fix this issue. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and Asterisk...not working...
Chuck, The first thing I would do is to upgrade the load to version 6 or higher. I'm running the latest...version 7.2. (I'm very happy with it) Are you using TFTP to load the configuration or manually configuring the 7960? I know it's a pain to setup TFTP just for a quick test. However, it's well worth it. If you have a CCO account you can find the latest load and config files here: http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 After getting the infrastructure in place the following link was all I needed to get my 7960 phones working properly: http://www.voip-info.org/wiki-Asterisk%20phone%20cisco%2079xx However, the 7960 does have some basic error logging. I'm not sitting in front of it right now so I can't tell you the key combinations. Hint: I went from version 3.2 like you to 7.2. However, as an interim step I had to go to 6.0 first. Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C Wegrzyn Sent: Saturday, September 25, 2004 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7960 and Asterisk...not working... Hi! I have Asterisk up and running and have KPhone, IAXtel and FWD connected to it and working. I have a Cisco 7960 with version 3 (App. Load ID POS3-2-00) software. I have configured the 7960 correctly, I think; I have set everything - name, shortname, auth.name and display name set to 200. I have set the password to 200. I've set the proxy address/port to 192.168.1.117/5060. I can't seem to get the phone to connect to Asterisk, though Kphone works fine. Does anyone have an idea of what I am doing wrong? TIA, Chuck Wegrzyn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and Asterisk...not working...
Is there a place to get the software load for the Cisco phone without having a support contract? Buying the phone was costly enough, but now needing to pay for the software to fix it is really poor! Chuck Wegrzyn Chad Brown wrote: Chuck, The first thing I would do is to upgrade the load to version 6 or higher. I'm running the latest...version 7.2. (I'm very happy with it) Are you using TFTP to load the configuration or manually configuring the 7960? I know it's a pain to setup TFTP just for a quick test. However, it's well worth it. If you have a CCO account you can find the latest load and config files here: http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 After getting the infrastructure in place the following link was all I needed to get my 7960 phones working properly: http://www.voip-info.org/wiki-Asterisk%20phone%20cisco%2079xx However, the 7960 does have some basic error logging. I'm not sitting in front of it right now so I can't tell you the key combinations. Hint: I went from version 3.2 like you to 7.2. However, as an interim step I had to go to 6.0 first. Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C Wegrzyn Sent: Saturday, September 25, 2004 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7960 and Asterisk...not working... Hi! I have Asterisk up and running and have KPhone, IAXtel and FWD connected to it and working. I have a Cisco 7960 with version 3 (App. Load ID POS3-2-00) software. I have configured the 7960 correctly, I think; I have set everything - name, shortname, auth.name and display name set to 200. I have set the password to 200. I've set the proxy address/port to 192.168.1.117/5060. I can't seem to get the phone to connect to Asterisk, though Kphone works fine. Does anyone have an idea of what I am doing wrong? TIA, Chuck Wegrzyn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ring delay
Hi people, I'm having some trouble with my analog phone on Zap/1 not ringing directly when i call it from the PSTN via Zap/4, after about 3 rings the analog phones rings. Now I understand that there is a slight delay of up to 3 rings where is the tone detection, so no phantom calls will get through. Anyway is there any way to disable this or try to tweak it, I want my analog to ring direct not after 3 tones from the caller. I have set usecallerid=no. Regards Fredrik vK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and Asterisk...not working...
C Wegrzyn wrote: Is there a place to get the software load for the Cisco phone without having a support contract? Buying the phone was costly enough, but now needing to pay for the software to fix it is really poor! That's the Cisco way!! They're not content to charge a premium price for their hardware; they make you pay for their bugfixes, too. In the ideal world companies that treat their customers this way would not be able to compete, but Cisco's de facto monopoly in the router market allows them to treat their customers as if they were their inmates. Not much choice until somebody else comes along with something as good or better for the same price, and makes them compete. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non-PRI T1 configuration
SF framing is called d4 in the zaptel.conf. And use ami instead of b8zs.
If you want those changed, it will be basically a new circuit from your
telco!
You say that you have it running into a channel bank now. What type of
channel units are in the channel bank? That will tell us what type of
signalling is on the channels. There is more to the signalling/chan type
than you have learned about yet.
I am not sure on DNIS. They may be just sending the callerid the same way
it's sent over a analog line. If it was my circuit and my pbx and migrating
to *, I would want to convert this to an ISDN PRI.
Lyle
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 1:01 PM
Subject: [Asterisk-Users] Non-PRI T1 configuration
I'm trying to hook up a non-PRI fractional T1 using a T400P port. The
Telco says that it is provisioned as AMI with SF (not ESF) and that they
are
signalling by sending down a straight DS1 (I'm not sure what exactly
that
means). They are also sending DNIS over these channels. I currently run
it
through a channel bank for my IVR application and it works fine but I am
now
trying to convert to *.
This leaves me with three questions. First, * does not have an option
for
SF framing. If I use ESF, should that work or is there another way?
Second, how do I configure the channel signalling in both zaptel.conf
and
zapata.conf?
Third, how can I capture the DNIS in this situation or will it
automatically be available in the ${EXTEN} variable and also passed to AGI
scripts?
I would appreciate any help.
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RE: [Asterisk-Users] Cisco PIX and Asterisk
It works fine for me. I have a handful of Cisco 7960s behind a PIX firewall and they register to a Asterisk server outside of the PIX with no trouble at all. I didnt do anything special to the PIX (i.e. no access list entries). The tricks I found to make it work generally apply to any setup where the clients are behind NAT. I also run the tftp server for the phones to get configs inside the firewall, and the SIPDefault.cnf file specifies the proxy address outside of the firewall. In the Cisco phone config I have these NAT settings: nat_enable: 1 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled And the sip.conf entry for this peer is: [7000] type=friend nat=yes qualify=yes context= secret= callerid= host=dynamic canreinvite=no dtmfmode=rfc2833 timer_register_expires: 120 Setting the registry timer to 120 seconds causes the phone to send out a packet at least every 2 minutes which will open a UDP xlate on the PIX for the session. Then the trick is to use both nat=yes and qualify=yes so Asterisk chats with the phone pretty often. The interval of OPTIONS or REGISTER messages between Asterisk and phone definitely needs to be shorter than the PIXs UDP xlate timeout or the PIX will close the xlate and you wont be able to pass packets into the phone for an incoming call. Note that you can put a numeric value after qualify= instead of yes to fine-tine the interval at which it sends a OPTIONS message. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: Saturday, September 25, 2004 8:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco PIX and Asterisk I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and Asterisk...not working...
Yes, that's tough. A couple things though... 1. To be fair...My 3.2 load did work against Asterisk. I just feel that troubleshooting should begin with the latest bug fixes applied if possible. 2. You may be able to contact Cisco technical support to get the latest firmware / files. Before I put a contract on one of my phones just for the purposes of downloading the latest loads I was able to convince one of the techs to provide me the latest load and supporting files for free. - It's worth a shot! Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Saturday, September 25, 2004 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 and Asterisk...not working... C Wegrzyn wrote: Is there a place to get the software load for the Cisco phone without having a support contract? Buying the phone was costly enough, but now needing to pay for the software to fix it is really poor! That's the Cisco way!! They're not content to charge a premium price for their hardware; they make you pay for their bugfixes, too. In the ideal world companies that treat their customers this way would not be able to compete, but Cisco's de facto monopoly in the router market allows them to treat their customers as if they were their inmates. Not much choice until somebody else comes along with something as good or better for the same price, and makes them compete. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Free G.729 ready for download
In article [EMAIL PROTECTED], Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 25 Sep 2004, Steve Underwood wrote: But the patches aren't a derived work. That is the value they have here. There are an independant adjunct work. According to most lawyers a patch _is_ a derived work in nearly all circumstances. E.g. a novel based on the characters from a novel by another author is a derived work. No, a patch *itself* is not a derived work. It is a set of instructions enabling the user of the patch to create a derived work from the original. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I dial one unbusy channel of 4 available?
Hi. I'm using asterisk as a PSTN - SIP gateway, so that you can call to any of the 4 PSTN lines connected to the asterisk box from and dial your number, and asterisk will dial out through one of the 4 sip accounts I have on a SIP - PSTN provider. I think of something like this in the extensions.conf [incoming] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,2 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,5 ; Set Response Timeout to 10 seconds exten = s,5,BackGround(welcome_and_dial_your_number) ; exten = _.,1,Dial(SIP/[EMAIL PROTECTED]) ;*** I dont know what to write instead of the line marked with ***. A multiple dial like following is not the solution I think. exten = _.,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]) How can I know the free (or busy, is the same to me) SIP channels at any moment? Is there any built-in var? Thanks in advance. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0439-2, 24/09/2004 Tested on: 25/09/2004 21:24:27 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway (CF based Aseterisk)
You could start buy downloading my .iso (29mb bootable ) and use that as a basis for your system. I've already modified it for a CF card based system. Essentially it depends what sort of interface to the PSTN you want. E1/T1 and analog should work fine with my cd - but I've not built it for use with CAPI or the QuadBRI cards... you can grab it at http://www.automated.it/asterisk/ It's not v1 of * but I am trying to find the time to update to a newer CVS version, however I will only do that once I'm happy running that particular version myself... HTH Andy Andy, I would be interested in a CF version too. Please, keep us posted on any progress. Thanks, Yiannis. What about Building as asterisk system based on a distribution like Bering-uClibc: http://leaf.sourceforge.net/mod.php?mod=userpagemenu=910page_id=36 It is really a firewall/router system, but has a ton of other packages available and it is easy to convert to running it on a CF card. I deploy my firewalls using this package on a CF card. They have lots of packages available as well: http://leaf.sourceforge.net/mod.php?mod=userpagemenu=91017page_id=51 It would be neat to have a Asterisk package that you could use for it. You would need to have access to a HDD to store voicemail since Bering uses a ram based file system. I wonder if running Asterisk from a ram file system would have benefits? Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and Asterisk...not working...
On Sat, 2004-09-25 at 13:48, Brian Capouch wrote: In the ideal world companies that treat their customers this way would not be able to compete, but Cisco's de facto monopoly in the router market allows them to treat their customers as if they were their inmates. Not much choice until somebody else comes along with something as good or better for the same price, and makes them compete. For most people that would be called Polycom Soundpoint IP 300, 500, and 600. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Sending to Cisco 3660 Sip Endpoint
All, I am trying to do a dial to a cisco3660 endpoint. see the below extensions.conf, sip.conf, and output to see my problem. Thanks in advance for any input. In the debug look for the WARNING lines. thanks! exten = 5149053538,1,Answer exten = 5149053538,2,Wait,2 exten = 5149053538,3,Playback(you-sound-cute) exten = 5149053538,4,Dial(SIP/[EMAIL PROTECTED],5) exten = 5149053538,105,Hangup [general] disallow=all allow=ulaw allow=alaw allow=g729 [melbourne] type=friend defaultip=xxx.xxx.xxx.xxx context=demo [montreal] type=friend context=demo defaultip=yyy.yyy.yyy.yyy *CLI Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E From: sip:[EMAIL PROTECTED];tag=F9E311A8-246C To: sip:[EMAIL PROTECTED] Date: Sat, 25 Sep 2004 19:51:18 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 436292417-241373657-3182559241-3907589232 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1096141878 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 194 v=0 o=CiscoSystemsSIP-GW-UserAgent 1631 5118 IN IP4 yyy.yyy.yyy.yyy s=SIP Call c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 19366 RTP/AVP 0 c=IN IP4 yyy.yyy.yyy.yyy a=rtpmap:0 PCMU/8000 a=ptime:20 20 headers, 9 lines Using latest request as basis request Sending to yyy.yyy.yyy.yyy : 5060 (non-NAT) Found RTP audio format 0 Peer audio RTP is at port yyy.yyy.yyy.yyy:19366 Found description format PCMU Capabilities: us - 0x10c(ULAW|ALAW|G729A), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Found no matching peer or user for 'yyy.yyy.yyy.yyy:58107' Looking for 5149053538 in default list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E From: sip:[EMAIL PROTECTED];tag=F9E311A8-246C To: sip:[EMAIL PROTECTED];tag=as2f5e7572 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to yyy.yyy.yyy.yyy:5060 -- Executing Answer(SIP/yyy.yyy.yyy.yyy-08141378, ) in new stack We're at xxx.xxx.xxx.xxx port 12034 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x8(ALAW) Answering with preferred capability 0x100(G729A) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E From: sip:[EMAIL PROTECTED];tag=F9E311A8-246C To: sip:[EMAIL PROTECTED];tag=as2f5e7572 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 210 v=0 o=root 19664 19664 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 12034 RTP/AVP 0 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - to yyy.yyy.yyy.yyy:5060 -- Executing Wait(SIP/yyy.yyy.yyy.yyy-08141378, 2) in new stack Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK1D34 From: sip:[EMAIL PROTECTED];tag=F9E311A8-246C To: sip:[EMAIL PROTECTED];tag=as2f5e7572 Date: Sat, 25 Sep 2004 19:51:18 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines -- Executing Playback(SIP/yyy.yyy.yyy.yyy-08141378, you-sound-cute) in new stack -- Playing 'you-sound-cute' (language 'en') -- Executing Dial(SIP/yyy.yyy.yyy.yyy-08141378, SIP/[EMAIL PROTECTED]|5) in new stack We're at xxx.xxx.xxx.xxx port 14742 Answering/Requesting with root capability 4 Answering with preferred capability 0x8(ALAW) Answering with preferred capability 0x100(G729A) Answering with non-codec capability 0x1(G723) 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:0 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd From: 8138174204 sip:[EMAIL PROTECTED];tag=as24022d46 To: sip:[EMAIL PROTECTED]:0 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 25 Sep 2004 19:47:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 266 v=0 o=root 19664 19664 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 14742 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to xxx.xxx.xxx.xxx:0 Sep 25 15:47:21 WARNING[1110272944]: chan_sip.c:598 __sip_xmit: sip_xmit of 0x81487dc (len 755) to xxx.xxx.xxx.xxx returned
RE: [Asterisk-Users] Cisco PIX and Asterisk
Thats Great news. Thanks for the information. What version of the PIX IOS you running? Do you have sip fixup protocol enabled? I have found a workaround, install onDo sip server on a machine behind the PIX. The phones register to that, on the pix port forward to the onDo sip server. But I would much rather get it working without having to do that. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hagler Sent: 25 September 2004 19:59 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco PIX and Asterisk It works fine for me. I have a handful of Cisco 7960s behind a PIX firewall and they register to a Asterisk server outside of the PIX with no trouble at all. I didnt do anything special to the PIX (i.e. no access list entries). The tricks I found to make it work generally apply to any setup where the clients are behind NAT. I also run the tftp server for the phones to get configs inside the firewall, and the SIPDefault.cnf file specifies the proxy address outside of the firewall. In the Cisco phone config I have these NAT settings: nat_enable: 1 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled And the sip.conf entry for this peer is: [7000] type=friend nat=yes qualify=yes context= secret= callerid= host=dynamic canreinvite=no dtmfmode=rfc2833 timer_register_expires: 120 Setting the registry timer to 120 seconds causes the phone to send out a packet at least every 2 minutes which will open a UDP xlate on the PIX for the session. Then the trick is to use both nat=yes and qualify=yes so Asterisk chats with the phone pretty often. The interval of OPTIONS or REGISTER messages between Asterisk and phone definitely needs to be shorter than the PIXs UDP xlate timeout or the PIX will close the xlate and you wont be able to pass packets into the phone for an incoming call. Note that you can put a numeric value after qualify= instead of yes to fine-tine the interval at which it sends a OPTIONS message. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: Saturday, September 25, 2004 8:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco PIX and Asterisk I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Application almost there..Dialplan challenges
Aloha, I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN. I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System. I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call is completed. I can not seem to get rid of the need to dial a lead digit. I don't need any other digits - i.e. voicemail, park - we aren't using any * 'features' just as a SIP-FXS gateway. Is it posible so I can create templates to collect the number and send the call to the Cisco when the template is completed 911 411 611 1[2-9]XX-XXX-XXX [2-9]XX- . The users are not likeing to have to dial '9' Looking forward to updateing to 1.0.0 Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote: Anyone else having the problems that Gary is reporting? Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel 2.6) and i had to add a linux 26 at the end of the make line, otherwise all kinds of weird things happened. Also, in /etc/init.d/zaptel, insmod doesn't work properly. It has to be replaced with modprobe. I have no idea why. There are some other changes i've made to the initialization scripts, to bring them closer to Red Hat best practices. I'll probably email you privately when i'm closer to a stable state. Anyway, the RPMs are way cool! :-) Thanks, -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reproducible problem with X100P... any suggestions?!
Hi Everyone, I've been playing around with Asterisk for awhile now, and keep having this intermittent problem with my X100P... Here is my setup: Linux Kernel 2.4.26 Wildcard TDM400P (One FXS port) Wildcard X100P (One FXO port) Running the 1.0 release of Asterisk and the Zaptel drivers It seems like whenever I run the server for a few hours with regular usage... my FXO port will get hung up on some random call. For example, here is a call that has been stuck for about 20 hours: lilith*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/1-1 (local 94252390158 2 ) Up Congestion (Empty) 1 active channel(s) lilith*CLI show channel zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1096069447.19 Caller ID: Main Extension 100 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor: 19 Frames in: 3727326 Frames out: 0 Time to Hangup: 0 Elapsed Time: 20h42m14s -- PBX -- Context: local Extension: 94252390158 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Congestion Data: (Empty) Stack: 0 Blocking in: ast_waitfor_nandfds Whenever this happens, if I try and dial out using the extension on my FXS port I get these messages from the server and a rapid busy signal: -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, Zap/1/2271229) in new stack Sep 25 13:08:18 NOTICE[573457]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Congestion(Zap/2-1, ) in new stack == Spawn extension (local, 92271229, 2) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' I can easily fix the problem by restarting the Asterisk server... however, this is obviously less than ideal. :-( So do any of you Asterisk guru's out there have a suggestion or two on how I can debug this problem? I've tried looking through the system logs and haven't found anything particularly helpful. Also, it doesn't seem to be correlated to the version of Asterisk that I'm running (I can reproduce it on RC1, RC2, CVS checkout, etc.) I'm starting to wonder if there is a hardware problem with my X100P... Anyway, any help or suggestions would be greatly appreciated!! Thanks, Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] German Termination and DIDs
Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Simple Manager Proxy
If you have developed CGI, PHP or other synchronous web-based applications that utilize the Asterisk manager interface, you know that they don't scale well, since each invocation from the web requires a connection to Asterisk and authentication there (thus putting a potentially large amount of connection and authentication load directly onto asterisk). There has been some discussion as to how to address this; some folks are talking about databases, others have developed specialized 'middleware' to bridge between the code at the edge and Asterisk. Nicolas Gudino's Flash Operator Panel is one such piece of code. I had a need for a much simpler proxy than his op_server.pl; to meet my need I re-worked and simplified his code. See below for this simplified proxy: http://www.popvox.com/simpleproxy.pl It's *very* simple: connects to Asterisk manager with a single authenticated connection, and listens on a configurable port (1234 by default) for inbound connections. Any commands passed from client-proxy are forwarded to Asterisk. Any events passed from asterisk-proxy are forwarded to all connected clients. They all share a common connection context so all clients will see the same thing, all clients will share the rights of the authenticated user. I make no pretense that this is particularly good code; I'm putting it out there for now as it helps me with testing something I'm working on, and it may be of use to others -- maybe we can start the ball rolling on something a bit more robust. Uses for this include: - Making a web-services/XMLRPC wrapper for asterisk manager - Building simple web-based applications - Backend for scalable, heterogeneous operator panels - Insulating Asterisk manager internals from user community Some potential next steps/enhancements for this basic design might be: - Test for robustness/IO interruptions on either side - Creating a connection pool of n (configurable) connections to * manager - Tracking connection contexts for clients - Redo with c/pthreads for speed (imapproxy is someplace to look) - Utilize libwrap to control access - Implement a simple authentication mechanism - Add TLS to clients for secure manager interactions Right now it is assumed that you will use this proxy in a secured environment -- either listening on localhost only, on a private LAN or behind a firewall. If you do not take some precautions you may be opening up a completely unauthenticated proxy to your * box!! Let me know if you have questions/thoughts/comments about this. Thanks, Dave -- David Troy popvox, llc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: CTI development
from asterisk' point of view holding onto some sort of,a dn obtainign some sort of uniq ID can be done easily via AGI and variableshowever, it sounds like, what you're talking about is more of an app (with several calls) and an resource too... maybe not really certain. --On Saturday, September 25, 2004 07:33 -0400 David Cook [EMAIL PROTECTED] wrote: Or what is it that you meant in particular? I'l bet he means 3rd party call control like in a traditional CTI deployment ala Cisco ICM, Genesys or an oldie-but-goodie, IBM CallPath DirectTalk. (Net-net version) Basically, a scratch-pad type area of ~2K that gets created/destroyed with every call and _follows_ the call for its life in the system. Olus the ability of a 3rd party computer application aka softphone to control the telephone appliation - this part we've got but still needs some modification for true CTI. (Example) So the caller gets to the IVR. The IVR pushes data relevant to the current call onto the scratch pad using a unique call event ID then xfers the call to the call centre Q. The call gets allocated to an agent in the Q. Their desktop application gets an alerting message which is basically a ring event alerting them that they are about to get the next event including the internal ID of the event. (In traditional environments this happens _slightly_ before the phone rings. The application then reads the scratch pad data associated with the call event ID so the desktop can have full context of what has gone before in the call. The desktop application then does whatever it needs to do in the customer environment - this is custom development - the CTI vendor offers an SDK for interface to their softphone product. The desktop application needs the ability to also write/update to the scratch pad as there may be a need to xfer the call to another agent or back to the IVR which should be able to read the updated data. I may not have the skill to code all of the application, but I'm a call centre solution architect. If anyone would like to bring this functionality to Asterisk I would be excited to offer industry advice. There are lots of gotchas in the CTI world that are completely _not_ related to programming skill. The wrong implementation simply won't have a market. dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Undocumented Features quote of the moment... It's not the one bullet with your name on it that you have to worry about; it's the twenty thousand-odd rounds labeled `occupant.' --Murphy's Laws of Combat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application almost there..Dialplan challenges
Matt, I am tring to use cisco as a sip to pstn gw as well. are you using an inbound sip dial-peer? or is not required? for inbound h323 calls its not but i keep getting Sep 25 15:47:23 WARNING[1087986608]: chan_sip.c:598 __sip_xmit: sip_xmit of 0x81487dc (len 755) to 210.50.7.213 returned -1: Invalid argument when i send a sip call to my cisco 3660. see my earlier post today. Matt Darnell wrote: Aloha, I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN. I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System. I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call is completed. I can not seem to get rid of the need to dial a lead digit. I don't need any other digits - i.e. voicemail, park - we aren't using any * 'features' just as a SIP-FXS gateway. Is it posible so I can create templates to collect the number and send the call to the Cisco when the template is completed 911 411 611 1[2-9]XX-XXX-XXX [2-9]XX- . The users are not likeing to have to dial '9' Looking forward to updateing to 1.0.0 Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Winter Senior Network Engineer Planet-Telecom, Inc. Tampa FL (813)901-5182 Office (813)864-3162 Direct (813)817-4204 Mobile (813)881-9762 Fax -- AIM: mobofool ICQ: 3563403 MSN:[EMAIL PROTECTED] Y!:vt_fool ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free G.729 ready for download
Regardless of whether or not you have licensed G.729 from SIPRO independently of Digium, the distribution of the codec, linked against Intel's proprietary IPP library, is clearly and totally in direct violation of the terms of the GPL. There is no room for argument on this issue. We are doing our best to mitigate the situation by removing the link to the illegal software from the list, and I ask that no more members on the list post any URL's to illegal software. Setting aside the extreme tackiness of using our own mailing list to post illegal software, I am nothing short of appalled to see an Asterisk user taking advantage of our hard work and then producing a product to circumvent the very revenue stream which makes it possible for us to do so and to offer the LEGAL licensing of G.729 to the community. Digium has worked hard to produce Free Software for building a phone system and we have released EVERYTHING we've done under GPL or other open source license, with the exception of the items we are not permitted to make available under that license. For G.729 specificially we have had to make a large investment to make that possible. The GPL DOES provide certain requirements for anyone distributing Asterisk code or derivative works, however and this unlicensed version is clearly in violation of those obligations. Why on earth would we try make the even larger investment for legal G.723.1 if people are just going to break the law and violate the GPL in order to save a few bucks? If you don't like G.729 because of the patent and licensing issues, then don't use it, but if you do want to use G.729, please use it legally, by purchasing the Digium licenses, not by breaking the terms of the GPL and putting yourself at risk of well established patents (especially if you are in a country which honors software patents, since the GPL passes that patent responsibility back to you as the end user). We are happy to make Asterisk available to the community and to continue to work hard to expand and develop the product further, but it also demands a certain level of discipline from the users at large. Before you download the free, illegal GPL-violation version of the G.729 codec, remember that in doing so you are directly jeopardizing the project at large and our ability to continue to provide these sorts of features. Mark On Sat, 25 Sep 2004, Steve Underwood wrote: Danny Zak wrote: Hello TELUX, could anybody post something more about being legaly correct using this codec and the corresponding royalty's. It is very difficult to be legally correct with this. The IP holders don't have simple programs for selling licences in small quantities. If you buy licences from Digium, they deal with the IP issues on a larger volume basis. Unless you want to deploy thousands of copies, I doubt you can find a sane legal arrangement for doing it. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German Termination and DIDs
Hi, if i understand german telco regulations right (even for a german that's not an easy task...) then a provider may not assign a DID to a non-local client. This would mean that a provider in Berlin may not assign a DID to a client in Munich. So, assigning german DIDs to foreign clients would not be legal at all. Yeeehahh, regulations rule! :-) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-09-25 um 22.32 schrieb Eric Jacksch: Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with dialing out with TDM400P
--On Saturday, September 25, 2004 23:28 +1000 James Bean [EMAIL PROTECTED] wrote: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but that's not important), I can't ring out, I just get a busy signal and nothing comes indications.conf -- you can adjust that and then set your country setting to the au countryi can't remember how to do the latter right now, but that should get you going in the right direction methinks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German Termination and DIDs
Try www.sipgate.de . They have DID numbers available in 14 cities in Germany. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 25 Sep 2004, Klaus-Peter Junghanns wrote: Hi, if i understand german telco regulations right (even for a german that's not an easy task...) then a provider may not assign a DID to a non-local client. This would mean that a provider in Berlin may not assign a DID to a client in Munich. So, assigning german DIDs to foreign clients would not be legal at all. Yeeehahh, regulations rule! :-) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-09-25 um 22.32 schrieb Eric Jacksch: Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-25%5Cf6f0534ca2fc4ddf99b1a9bdad8698bcC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFC/R2
Hi, Steve Underwood wrote: Asterisk. I have been building and testing with the current * CVS code. I still need to work through the national variants, and get some of the them better tested. If you have the equipment ready to try MFC/R2 please tell me how you get on. I might have an R2 line this week for testing in Brazil. I would like to test it with 1.0.0 ? Any problem ? Does it have to be with CVS ? Great job! Thanks, Leonardo -- Leonardo Gomes Figueira [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue and Agent functionality
Chris,
I agree with your assessment of asterisk's queues. I took Robert's reply to
my original post, and came up with a way to tackle your first scenario (no
agents in queue=caller in limbo) with his idea of setting variables. My idea
deals with setting global variable states for each agent. I only have 4
agents, so it should work for me fairly easily. In the extensions.conf file
I would have something like this:
[globals]
GCSR1=off
GCSR2=off
GCSR3=off
GCSR4=off
Then, in the context where my agents log in/out of queue, I set the global
variable to on/off depending on their action. When the agent dials 800,
GCSR1 becomes 'on'. When they dial 801##, GCSR1 becomes 'off'.
[fromcsr1]
exten= 800,1,AgentCallbackLogin(101|[EMAIL PROTECTED])
exten= 800,2,SetGlobalVar(GCSR1=on)
exten= 800,3,Hangup
exten= 801,1,AgentCallBackLogin(101)
exten= 801,2,SetGlobalVar(GCSR1=off)
exten= 801,3,Hangup
Then, in my queue, I check for the value of GCSR1 before dumping them to the
queue. Otherwise, dump them to VM. Obviously, the GotoIf would have to check
if GCSR1 = on | GCSR2 = on | GCSR3 = on | etc... For my testing, I was just
using GCSR1.
[queue]
exten = 1,1,DigitTimeout,1
exten = 1,2,ResponseTimeout,1
exten = 1,3,GotoIf($[${GCSR1} = on]?4:5)
exten = 1,4,Queue(order|tT)
exten = 1,5,Goto(generalvm|s|1)
While this idea seems to make sense (in my head), I am unable to make it
work. For example, my GotoIF command does work, so the value of GCSR1 will
determine which path the caller takes. The part that doesn't work is in the
[fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect,
therefore, making my solution not work.
Does anyone have any ideas?
Thanks,
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Icide
Sent: Saturday, September 25, 2004 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Queue and Agent functionality
I've seen alot of posts lately on Queue and Agent functionality, and alot of
hacks to make them do different things that most call center managers want.
In the sake of doing this one time, I'd like to develop a single list of
request so we can consolidate a feature request for the Queue/Agent system.
Here are the ones that I run into the most:
1. Queue should know the status of agents assigned to a queue and act
accordingly.
Here are a couple examples of the problem.
A queue has no agents logged in and handling the queue, a call comes in for
the queue, the call remains in the queue until either an agent logs in, or
the queue reaches it's timeout. What it should do is immediately time out
setting priority +101. Normal timeout (caller in busy queue with agents
active) should exit with priority set +1.
A Queue has active agents in a prioritized fashion. Agent 1 is priority 1,
2 is 2, 3 is 3, and 4 is 4. Agent 1 needs to make an outbound call as does
agent 2. Both are now 'busy'. The Queue still attempts to call agent 1,
gets 'busy' back from the sip device (i've only tried this with sip), and
then the system appears to wait for something like 7-8 seconds before trying
the next agent in line.
2. The queue system should allow a set of messages to be played at specific
times. For example, a message that is played upon entry into the queue and
no other time, the current set of messages played every frequency=XX, a
message played to the caller when the call is accepted by an agent (eg
transfering), finally, a set of messages played to the user based upon a
predefined period int he config file.. see example below
message1-time=time in seconds
message1-frequency=never|once|always
message1=message1-file-loc
message2-time=time in seconds
message2-frequency=never|once|always
message2=message2-file-loc
Where a message messageX-file-loc is played never|once|always every
time in seconds.
if time is set to 0, or freqency is set to never, the message is not played.
If time is set to 0, and frequency is set to once, message is played at
messagex-time, and never again.
if time is set to 0 and frequency is set to always, message is played every
messagex-time in seconds.
3. Agent timeout (logs the agent off if they do not respond to a ring in a
defined about of time) does not track across calls. For example, if an
agent steps away and forgets to log out, then thier phone will ring based
upon whatever call strategy is used. If the agent timeout is set higher
than the time the queue polls a set of agents they will never be logged out.
The timer needs to increment per agent across multiple polls. So if my
queue poll timer is 20 secons, but the agent timeout is set to 60 seconds,
the preferred function would be to log the agent out of the queue if they
completely miss three poll events.
4. If a caller empties a handled queue (active agents) with no callers, the
caller will still hear messages (you are first in queue, etc.). This should
not occur. Someone posted a 2-line patch on
[Asterisk-Users] Dropping numbers on dialout through tdm400p
Specs
FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO
Standard analogue handset plugged in with pstn line.
Problem:
When I go to dialout it drops numbers on the outgoing number.
Keys dialed from handset were
9 0418800185
I tried hitting the keys slowly as well as at my normal speed, all tones
are heard in the handset for all numbers.
Error in asterisk -vvvgc
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/088008) in new stack
-- Called g2/088008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
== Spawn extension (internal, 9088008, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/0488008) in new stack
-- Called g2/0488008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
== Spawn extension (internal, 90488008, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'
/etc/zaptel.conf
fxols=1
fxsls=4
Loadzone=au
/etc/zapata.conf
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690 ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above
exten = s,3,Hangup
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
exten = _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9X.,2,Congestion()
--
Secondary issue, when an incoming call into the asterisk box arrives on
the asterisk terminal it shows callerid of the caller as 690 which is
the extension number that rings not the actual other persons caller id.
James
___
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RE: [Asterisk-Users] Queue and Agent functionality
Hi! [fromcsr1] exten= 800,1,AgentCallbackLogin(101|[EMAIL PROTECTED]) exten= 800,2,SetGlobalVar(GCSR1=on) exten= 800,3,Hangup determine which path the caller takes. The part that doesn't work is in the [fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect, therefore, making my solution not work. Does anyone have any ideas? Unfortunately AgentCallbackLogin() _itself_ initiates the hangup, which means that any following priorities in your dialplan are useless. Besides your approach isn't yet perfect, what if an agent gets auto-logged out because he/she hasn't answered within the time limit? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue and Agent functionality
Here is my resolution to the problem, I use AgentLogin vs
AgentCallBackLogin.
This is a long post, but I think it is very useful... :)
Call comes in via DID, queueable is a macro I wrote. ty_voice and
voice are
two sound files. The first one is used to play a Thank you for calling
XXX.
The second is what the agent will here so they know what number was
dialed.
//Agents Dial 7100 to login
exten = 7100,1,Answer()
exten = 7100,2,AGI,SetAgent.agi
exten = 7100,3,AgentLogin(${AgentID})
//My AGI just gets the AgentID, verifies it exists and
//then adds it to the queue. This is done through an
//AGI because I add every agent to multiple queues
//
// I then have a script in my timeclock programming that when
// our employees Clock out they are removed from all queues.
//
// I did have it set so that when the employee clocked in, they were
// added to the queues automatically, but this causes problems when
// they are supposed to clock in but not get on the phones right away.
#!/usr/bin/perl
use Asterisk::AGI;
use Asterisk::Manager;
$AGI = new Asterisk::AGI;
open (F, /etc/asterisk/agents.conf);
while (F) {
if ($_ =~ /^agent = (\d*),(\d*),(.*)/) {
$Agent{$1} = 1;
}
}
close(F);
my %input = $AGI-ReadParse();
$ID = $AGI-get_data(agent-user, 3000, 3);
$ID =~ s/#//g;
while ((!$Agent{$ID}) ($Count 5)) {
$AGI-stream_file(agent-incorrect);
$ID = $AGI-get_data(agent-user, 3000, 3);
$ID =~ s/#//g;
$Count++;
}
if (!$Agent{$ID}) {
$AGI-stream_file(agent-incorrect);
exit;
}
$AGI-set_variable('AgentID', $ID);
$AGI-exec('AddQueueMember', 'PlaceOrders|Agent/'.$ID);
// This sets up for incoming calls, and passing them to a queue
exten = 1022,1,Macro(queueable,ty_voice, voice)
[macro-queueable]
exten = s,1,answer
exten = s,2,Wait(2)
exten = s,3,Playback(${ARG1})
exten = s,4,SetVar(Announce=${ARG2})
exten = s,5,Goto(MainMenu|s|1)
[MainMenu]
All s, extensions are used to play the menu
exten = s,1,Playback(MayBeRecorded)
exten = s,2,BackGround(Orders)
exten = s,3,BackGround(digits/1)
...
exten = s,18,WaitExten(15)
exten = s,19,Goto,2
# If 1 is pressed
exten = 1,1,PlayBack,hold_pcs # Play a please hold message
exten = 1,2,Queue(PlaceOrders|t||${Announce})
exten = 1,3,Goto(9|2)
On Saturday, September 25, 2004, at 05:43 PM, Marco Nicolayevsky wrote:
Chris,
I agree with your assessment of asterisk's queues. I took Robert's
reply to
my original post, and came up with a way to tackle your first scenario
(no
agents in queue=caller in limbo) with his idea of setting variables.
My idea
deals with setting global variable states for each agent. I only have 4
agents, so it should work for me fairly easily. In the extensions.conf
file
I would have something like this:
[globals]
GCSR1=off
GCSR2=off
GCSR3=off
GCSR4=off
Then, in the context where my agents log in/out of queue, I set the
global
variable to on/off depending on their action. When the agent dials 800,
GCSR1 becomes 'on'. When they dial 801##, GCSR1 becomes 'off'.
[fromcsr1]
exten= 800,1,AgentCallbackLogin(101|[EMAIL PROTECTED])
exten= 800,2,SetGlobalVar(GCSR1=on)
exten= 800,3,Hangup
exten= 801,1,AgentCallBackLogin(101)
exten= 801,2,SetGlobalVar(GCSR1=off)
exten= 801,3,Hangup
Then, in my queue, I check for the value of GCSR1 before dumping them
to the
queue. Otherwise, dump them to VM. Obviously, the GotoIf would have to
check
if GCSR1 = on | GCSR2 = on | GCSR3 = on | etc... For my testing, I was
just
using GCSR1.
[queue]
exten = 1,1,DigitTimeout,1
exten = 1,2,ResponseTimeout,1
exten = 1,3,GotoIf($[${GCSR1} = on]?4:5)
exten = 1,4,Queue(order|tT)
exten = 1,5,Goto(generalvm|s|1)
While this idea seems to make sense (in my head), I am unable to make
it
work. For example, my GotoIF command does work, so the value of GCSR1
will
determine which path the caller takes. The part that doesn't work is
in the
[fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect,
therefore, making my solution not work.
Does anyone have any ideas?
Thanks,
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Icide
Sent: Saturday, September 25, 2004 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Queue and Agent functionality
I've seen alot of posts lately on Queue and Agent functionality, and
alot of
hacks to make them do different things that most call center managers
want.
In the sake of doing this one time, I'd like to develop a single list
of
request so we can consolidate a feature request for the Queue/Agent
system.
Here are the ones that I run into the most:
1. Queue should know the status of agents assigned to a queue and act
accordingly.
Here are a couple examples of the problem.
A queue has no agents logged in and handling the queue, a call comes
in for
the queue, the call remains in the queue until either an agent logs
in, or
the queue reaches it's timeout. What it should do is immediately time
out
setting priority +101. Normal timeout (caller
Re: [Asterisk-Users] German Termination and DIDs
Klaus-Peter Junghanns schrieb: Hi, if i understand german telco regulations right (even for a german that's not an easy task...) then a provider may not assign a DID to a non-local Hi, it's right, that german RegTP, the authority, who assigns number ranges to telcos, now explicitely forbids to do so. The reason is not an explicit regulation right, but two german telcos (DTAG and Arcor), which consider local telephone numbers used by a non-local subscribers as spoofing. An alternative are personal or service numbers or maybe in some months 032- numbers reserved for voip. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Free G.729 ready for download
Hi All, I consider the License fee charged by digium for G.729 as very reasonable, and hope people agree and do nothing to jeopardize this project. Right now I don't use G.729 at all, however if and when I do, I have no reason to seek an alternative to what Digium provides. At the very least I would be confident that I am in no way breaking the law, and have the satisfaction of have contributed back to the product, be it in a very small way. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer Sent: 25 September 2004 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Zak Subject: Re: [Asterisk-Users] Free G.729 ready for download Regardless of whether or not you have licensed G.729 from SIPRO independently of Digium, the distribution of the codec, linked against Intel's proprietary IPP library, is clearly and totally in direct violation of the terms of the GPL. There is no room for argument on this issue. We are doing our best to mitigate the situation by removing the link to the illegal software from the list, and I ask that no more members on the list post any URL's to illegal software. Setting aside the extreme tackiness of using our own mailing list to post illegal software, I am nothing short of appalled to see an Asterisk user taking advantage of our hard work and then producing a product to circumvent the very revenue stream which makes it possible for us to do so and to offer the LEGAL licensing of G.729 to the community. Digium has worked hard to produce Free Software for building a phone system and we have released EVERYTHING we've done under GPL or other open source license, with the exception of the items we are not permitted to make available under that license. For G.729 specificially we have had to make a large investment to make that possible. The GPL DOES provide certain requirements for anyone distributing Asterisk code or derivative works, however and this unlicensed version is clearly in violation of those obligations. Why on earth would we try make the even larger investment for legal G.723.1 if people are just going to break the law and violate the GPL in order to save a few bucks? If you don't like G.729 because of the patent and licensing issues, then don't use it, but if you do want to use G.729, please use it legally, by purchasing the Digium licenses, not by breaking the terms of the GPL and putting yourself at risk of well established patents (especially if you are in a country which honors software patents, since the GPL passes that patent responsibility back to you as the end user). We are happy to make Asterisk available to the community and to continue to work hard to expand and develop the product further, but it also demands a certain level of discipline from the users at large. Before you download the free, illegal GPL-violation version of the G.729 codec, remember that in doing so you are directly jeopardizing the project at large and our ability to continue to provide these sorts of features. Mark On Sat, 25 Sep 2004, Steve Underwood wrote: Danny Zak wrote: Hello TELUX, could anybody post something more about being legaly correct using this codec and the corresponding royalty's. It is very difficult to be legally correct with this. The IP holders don't have simple programs for selling licences in small quantities. If you buy licences from Digium, they deal with the IP issues on a larger volume basis. Unless you want to deploy thousands of copies, I doubt you can find a sane legal arrangement for doing it. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * works, but after a few seconds audio always stops.
Using X-Lite FWD soft phone, I can register, get to the 'demo' menu extension, but that's it. Audio starts, then after a few seconds stops, with packets still being passed. Anyoen have any clues? Yes there are firewalls between here and there, yes there is NAT at my end...What ports need punching, is rfc2833 the correct settign or should I use inband or info? TIA, I just cant' seem to find anything on the WIKI about it just sorta locking up like this. BTW if i hold/unhold the extension the audio will come back for a few more seconds. *CLI show version Asterisk CVS-05/31/04-22:00:51 built by [EMAIL PROTECTED] on a i686 running Linux *CLI using ztdummy for timing. -- Undocumented Features quote of the moment... It's not the one bullet with your name on it that you have to worry about; it's the twenty thousand-odd rounds labeled `occupant.' --Murphy's Laws of Combat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German Termination and DIDs
Try sipgate.de. They have free DIDs in many german citys and their rate into Germany is very affordable (aprx. $0.02 / min.) Their website is in German only though. Alfred. Klaus-Peter Junghanns wrote: Hi, if i understand german telco regulations right (even for a german that's not an easy task...) then a provider may not assign a DID to a non-local client. This would mean that a provider in Berlin may not assign a DID to a client in Munich. So, assigning german DIDs to foreign clients would not be legal at all. Yeeehahh, regulations rule! :-) best regards Klaus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
