[Asterisk-Users] Anyone knows how to receive a SIP call without registering gateway?
Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP. Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation. Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox SIP way which is: client registers to my proxy. Guess what, I can't find any samples on this!!, Can anyone please help?, I will probably need a sample sip.conf. and then, to make a test call, I can use another asterisk box and try asterisk to asterisk sip calls (without register) via the cli prompt. But I have no idea and I am intrigued. Thanks CS ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] slight echo via sip provider
Hi, Damon Estep wrote: Here is the setup; analog phone Linksys ata asterisk sip provider sonus GSX 9000 PSTN called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight echo of their voice. All of the Zapata.conf echotraining, echocancel, etc do not seem to apply here as there is no zap channel involved in the call. Correct. I assume that since the echo is toward the called party who is on the other side of the provider sonus softswitch and somewhere on the PSTN, that the echo is really coming from the providers media gateway/softswitch. This is possible, but not really likely. Most decent service providers use digital equipment and would (should) not introduct additional echo on their end. However, it is very well possible that your Linksys ATA and the connected analog phone are causing the echo. I'm not sure about the capabilities of the Linksys, but with Sipura's you can modify the line impedance settings to best match your equipment. Look for the Regional Tab at the top. There is a setting called FXS Port Impedance. Try various options in there - they should match your phone. Best regards, Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone knows how to receive a SIP call without registering gateway?
What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static IP that these calls will be coming from and only accept calls from that IP, but that's still not quite as secure as digest authentication that would be available via registration. If you know what IP the calls are coming from, you simply insert a host=XX.XX.XX.XX instead of host=dynamic in your sip.conf for that peer and calls should then come in as they did before without them having to register. If they are pre-pending digits on to the front of what you're interpreting as the dialed number/extension, you may choose to lop them off in extensions.conf, but aside from that this is fairly straight forward. On 9/14/05, C. Savinovich [EMAIL PROTECTED] wrote: Hello everyone, I am pulling my hair here because a carrier threw me curve early today.They want to send calls to my asterisk server using SIP.Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation.Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox SIP way which is: client registers to my proxy. Guess what, I can't find any samples on this!!, Can anyone please help?, I will probably need a sample sip.conf. and then, to make a test call, I can use another asterisk box and try asterisk to asterisk sip calls (without register) via the cli prompt. But I have no idea and I am intrigued.ThanksCS ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!
I suppose the question is now whether you would recommend buying one later, PaulH - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 13, 2005 5:22 PM Subject: Re: [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD! An update on this... I was wrong. The wireless problem was an altogether different issue. the wj0011 firmware finally made my phone useable, after 6 months of problems. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Callerid fails in any release after beta1 fails
On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote: I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. I am having the identical problem. I use the CVSHEAD Asterisk and do an update every couple of weeks or so. I did one last week and the caller id quit working on my two lines that have x100p cards. I didn't make any changes to my configuration files at that time, simply updated Asterisk. In the meantime I checked my configuration files carefully and don't see anything wrong. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 box single Asterisk
Brave is the person that wants to use 3 Fritz cards in one box Go with the Jurgens 8 bri or 2 quad Brior bri-e1 chan bank... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Tuesday, 13 September 2005 6:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 2 box single Asterisk Here's my suggestion. Do a dialplan thing where when all trunks on boxA are busy, they are sent via IAX to boxB which sends them out via the ISDN trunks... this way boxA will be your primary box and boxB is your spare box that takes over if everything else is busy... On Tuesday 13 September 2005 10:00, Asterisk Sales wrote: hello list, i need to setup an asterisk system with 5 ISDN trunks. i found C4 cards but they are very expensive. i found that if i use 5 AVM Fritz! cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. and i want, this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are using a panasonic PBX system but it is driving us crazy. we want to keep the existing pbx setup and add asterisk with it to handle the call center operations. we also need to communicate with pbx users from Asterisk. our pbx has 6 analog trunks. so we can use TDM400P please help how can i solve this situation will low cost and performance. best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.23/99 - Release Date: 12/09/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.24/101 - Release Date: 13/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Cards in Australia
Agreed - ATP are always good to deal with. PaulH - Original Message - From: Callum McGillivray [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 14, 2005 12:41 PM Subject: Re: [Asterisk-Users] Digium Cards in Australia Hi Rudolf, Talk to Australian Technology Partnerships (www.atp.org.au). Cheers, Callum [EMAIL PROTECTED] wrote: Hi, all Where can I get Asterisk Developer's PCI Kit in Australia? It is a TDM400P with 1FXS and 1 FXO module. I amight need an extra FXS module as well. Thanks, Rudolf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting call minutes on a GSM SIM
On Tue, 13 Sep 2005, trixter http://www.0xdecafbad.com wrote: On Wed, 2005-09-14 at 07:01 +0200, Remco Barende wrote: Hi! I'm considering to buy a GSM bridge to save on GSM calls. Right now they are offering subscriptions with 200 minutes each month for almost nothing, however the 400 minutes subscriptions are considerably more expensive. Most GSM bridges can cater for 2 SIM cards, is there a way for Asterisk to run the first SIM card to it's max and then switch to the second? (If one call would overlap I wouldn't mind). Asterisk would have to keep track of the minutes called each month for a SIM (channel?). On most bridges you can select the SIM you want by a dial prefix. I do not know about the specifics, but it seems to me that you would need an AGI that would track the usage and compare that before placing a call. To switch I do not know how you tell the sim adapter which one to use, but surely there must be a command somewhere, the mere fact that agi allows you to script something like this fairly easily means that it shouldnt be a big problem, assuming you code :) And you can even pick your favourite language given how the AGI talks to asterisk even 'unsupported' languages can be used. Thanks for the tip. I was actually thinking in the direction of putting the asterisk calling card application to use. I've never used it and wonder if it is at all possible to use it from within the dial plan instead of normally from an extension. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting call minutes on a GSM SIM
On Wed, 2005-09-14 at 09:30 +0200, Remco Barende wrote: Thanks for the tip. I was actually thinking in the direction of putting the asterisk calling card application to use. I've never used it and wonder if it is at all possible to use it from within the dial plan instead of normally from an extension. Yup. I will try to make it simple for the archives, or anyone else that is interested in doing this type of thing. You appear to know most of this already, but then again you arent the only person on this list :) Call the AGI from the dialplan when you want to. exten = 31337,1,answer exten = 31337,2,playback(welcome) exten = 31337,3,agi(blah.pl) replace blah.pl with whatever the name is, so long as its executable. blah.php blah a.out etc see asterisk.conf for where to place the agis astagidir = /some/path/to/asterisk/agi-bin -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pri release cause code mismatch
Hi! My asterisk (1.0.7) is connected to a Nortel pbx with Digium E100P card, both side are ETSI EuroISDN. I would like to reject an incomming call with cause code 34, but the Nortel PBX gets the value of 31 instead of 34. It seems to work on the asterisk side: Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 17162/0x430A) (Originator) Message type: SETUP (5) ... Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 49930/0xC30A) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 a2] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (2) ] My macro looks like: exten = s,1,SetVar(PRI_CAUSE=34) exten = s,2,Hangup According to the debug on Nortel it gets 31 cause code in the release complete q.931 message. Do you have any idea? Thanks, Miklos ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] callfile: How to invoke SetCallerPres ?
Probably easiest to set a variable to the number to be called and then jump to an extension to do whatever you want to do? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruno Voigt Sent: 13 September 2005 23:37 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] callfile: How to invoke SetCallerPres ? Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 and Asterisk: Calls always hang up
Hi :) I hope someone can help me (google cannot): My little asterisk receives calls via h323 from PSTN. I connected a Sipura phone to my asterisk. oh323 is installed and calls go into the right context but immediately after the phone is picked up a hangup is signalled and the call ends :( This is what I get: Inbound H.323 call 'ip$213.30.225.5:42873/1893' detected. Channel OH323/[EMAIL PROTECTED] created and attached for inbound H.323 call 'ip$213.30.225.5:42873/1893'. -- Executing NoOp(OH323/[EMAIL PROTECTED], h323 Call an 4999663-99!) in new stack -- Executing Playback(OH323/[EMAIL PROTECTED], tt-monkeysintro) in new stack Channel OH323/[EMAIL PROTECTED] answered. -- Playing 'tt-monkeysintro' (language 'en') Call 'ip$213.30.225.5:42873/1893' cleared. -- H.323 call 'ip$213.30.225.5:42873/1893' cleared, reason 24 (Call ended with Q.931 cause) Sep 14 10:30:42 WARNING[14895]: file.c:970 ast_waitstream: Unexpected control subclass '5' Call 'ip$213.30.225.5:42873/1893' with owner has already been cleared (2). Call 'ip$213.30.225.5:42873/1893' has been hungup. -- Hungup 'OH323/[EMAIL PROTECTED]' Call 'ip$213.30.225.5:42873/1893' without owner has already been cleared (2). Any ideas? Thanks and kind regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call restrictions
Hello, I want to use call restriction option. For example, there are 3 registered numbers that 100,200 and 300.I want 100 to call 200 but not 300, btw 300 can call both 100 and 200. How can i configure this? Thanks. Erdem HAKI ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetCIDName question
finally I did it - I put some of the vars in (double)quotes - this didn't work even if there's a space inside, the vars need not to be kept inside (double)quotes... You probably want to use 'database put' for changing incoming CID http://voip-info.org/tiki-index.php?page=database%20put *CLI database put cidname 111222 test user Updated database successfully *CLI database show cidname /cidname/111222 : test user so now when someone calls from 111.222., it will change the CID info to 'test user' On Tue, 2005-09-13 at 07:46, [EMAIL PROTECTED] wrote: Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from CID withheld to abc CID withheld If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with SetCIDName. At least it's not displayed on my cisco7960 with chan_sccp any suggestions what I've could have done wrong ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call restrictions
you really should read about the concept of a context in extension.conf, that will answer your question and is also a basic key to understanding Asterisk. http://www.voip-info.org is your friend. Christoph On Wednesday 14 September 2005 10:47, Erdem HAKİ wrote: Hello, I want to use call restriction option. For example, there are 3 registered numbers that 100,200 and 300.I want 100 to call 200 but not 300, btw 300 can call both 100 and 200. How can i configure this? Thanks. Erdem HAKI ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid fails in any release after beta1 fails
Richard Kashdan wrote: On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote: I am having the identical problem. I use the CVSHEAD Asterisk and do an update every couple of weeks or so. I did one last week and the caller id quit working on my two lines that have x100p cards. I didn't make any changes to my configuration files at that time, simply updated Asterisk. In the meantime I checked my configuration files carefully and don't see anything wrong. Callerid has stoped working for us as well from the SIP phones to the PRI. PRI to the SIP phones work fine. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway?
Well, a SIP authorization does not require a registration (in fact, registration should be primarily used to inform a registrar about thewhereabouts of a UA with dynamic IP address in order to handle incoming calls_for_ that UA). CS can just createfor his Asteriska "type=user" entry in sip.conf containing "username" (equal to the section's title) and "secret" both matching the remote peer's own: his Asterisk will then react toan INVITEfrom that peer with a "401" replycontaining a nonce as challenge; the peer will then retry the INVITE withvalid credentials based on the shared secret and the nonce. Enzo - Original Message - From: BJ Weschke To: C. Savinovich ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, September 14, 2005 2:49 PM Subject: Re: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway? What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static IP that these calls will be coming from and only accept calls from that IP, but that's still not quite as secure as digest authentication that would be available via registration. If you know what IP the calls are coming from, you simply insert a host=XX.XX.XX.XX instead of host=dynamic in your sip.conf for that peer and calls should then come in as they did before without them having to register. If they are pre-pending digits on to the front of what you're interpreting as the dialed number/extension, you may choose to lop them off in extensions.conf, but aside from that this is fairly straight forward. On 9/14/05, C. Savinovich [EMAIL PROTECTED] wrote: Hello everyone, I am pulling my hair here because a carrier threw me curve early today.They want to send calls to my asterisk server using SIP.Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation.Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox SIP way which is: client registers to my proxy. Guess what, I can't find any samples on this!!, Can anyone please help?, I will probably need a sample sip.conf. and then, to make a test call, I can use another asterisk box and try asterisk to asterisk sip calls (without register) via the cli prompt. But I have no idea and I am intrigued.ThanksCS___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sometimes dtmf passed, sometimes not (cisco 7960 SIP)
Just to answer my own query, I needed to set the devices to dtmfmode=inband in my sip.conf, and on the 7960 set Sip configuration - Out of Band DTMF - none The benefits of a good nights sleep :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mat Stace, Colewood Sent: 13 September 2005 22:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] sometimes dtmf passed,sometimes not (cisco 7960 SIP) [major snippage] I hope the above makes some sense, it's basically is it an asterisk or 7960 setting to make it pass dtmf whilst on a call Cheers (and apologies for semi-coherance) Mat -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.24/101 - Release Date: 13/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T.38 ATA
Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that from Sipura release notes for firmwares. Anything else (other then Cisco routers with FXS modules) with T.38, or at least can someone confirm me that Sipura SPA-2100 has T.38 (firmware version would be nice info also) ? Thank you very much. Nenad Radosavljevic ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf Syntax to match first digits
answer: SetCIDNum(0${CALLERIDNUM:2:20})
shows 20 digits of the number but strips the first 2, additionally a 0 is added
at the beginning.
yes, it is basic - but is it thoroughly documented somewhere? i'm sure that
there are lots of other syntax possibilities...
On Sat, 10 Sep 2005 11:58:32 +0200
ChB [EMAIL PROTECTED] wrote:
i'm sorry, i still have a problem to edit the callerID - since stripMSD
and prefix seem to work only for extensions, how can i edit the number
automated(i mean not like SetCIDNum(0650123123) but a valid rule for
all numbers beginning with 43)?
thanks for your input
christian
On Sat, 10 Sep 2005 09:46:21 +0200
ChB [EMAIL PROTECTED] wrote:
how does a GotoIf-challenge look like to match e.g. only the first two
digits? i want to strip the first two digits from an incoming pstn-call and
add a zero instead so when i forward a call to a mobile the called party
gets the correct number of the caller. at the moment, incoming calls from
the austrian pstn are recognized as e.g. 43650123123 by asterisk, when i
forward the call e.g. to a mobile, the austrian telcos add a +43 to the
number so it appears as +4343650123123 to the called person(when the first
digit would be a zero, it is beeing stripped and the +43 added so it would
appear correct). since not all calls should be handled that way, i need a
gotoif-challenge. but how does the challenge look like to match only the
first two digits? Thank you for your help!
regards
christian
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RE: [Asterisk-Users] NAT and SIP.conf update.
On Tue, 2005-09-13 at 09:31 -0700, canuck15 wrote: I don't recommend anyone use free dyndns via router support. If you reboot your router more than once or twice in a month or have a power outage or whatever dyndns stops updating the IP automatically and will cancel your account for too much activity. You won't know it for a few weeks until they send you an email saying your account will expire in a week unless you go to the site and ask them nicely to reset it. Not a big deal to do that but it becomes annoying when it keeps happening over and over. I have used their system for years, both free and paid, and have _never ever_ had that type of experience with the 20 or so systems I've got running. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Wrapup time for agents.
Hi, QueueMetrics version 0.9.5 rc 2, out today, does the trick and allows agent pause monitoring (together with the rest of the stuff). See http://queuemetrics.loway.it Thanks l. In data Wed, 14 Sep 2005 07:28:51 +0200, Callum McGillivray [EMAIL PROTECTED] ha scritto: Hey Kevin, That's pretty much what I was looking for - now the killer question... is there a way for me to monitor the total amount of paused time for each agent ? Essentially, I want to give agents the ability to wrap up calls according to their needs, but I also want a team leader to police it and make sure they are not using inordinate amounts of time. Cheers, Callum Kevin P. Fleming wrote: Alexander Lopez wrote: Agents logging out is the prefered method of saying I can't be bothered right now CVS HEAD also supports pause/unpause for agents, which allows them to be unavailable without the queue losing its statistics. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf Syntax to match first digits
take a look into the wiki... http://www.voip-info.org/wiki-Asterisk+variables ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura Registration time out, no incoming calls
Hi everybody, My Sipura device registers on an Asterisk server and works fine. Its default registration time out value is 3600s. But I've noticed that once in a while it stops receiving calls but dial out works fine. To solve this problem I've to change registration time out value to 10s. Why is it like that, why doesn't everything work fine with timeout value of 3600s? Zeeshan A Zakaria ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura Registration time out, no incoming calls
It's very possible your firewall is closing the connection. When you try to make a call it forces the phone to re-register. Are you using STUN? On 9/14/05, Zeeshan [EMAIL PROTECTED] wrote: Hi everybody, My Sipura device registers on an Asterisk server and works fine. Its default registration time out value is 3600s. But I've noticed that once in a while it stops receiving calls but dial out works fine. To solve this problem I've to change registration time out value to 10s. Why is it like that, why doesn't everything work fine with timeout value of 3600s? Zeeshan A Zakaria ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 ATA
Nenad Radosavljevic wrote: Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that from Sipura release notes for firmwares. Anything else (other then Cisco routers with FXS modules) with T.38, or at least can someone confirm me that Sipura SPA-2100 has T.38 (firmware version would be nice info also) ? The newest 2100 firmware has T.38. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 ATA
Can anyone recommend me ATA device that REALLY has T.38 built in. While I have not tested it myself (one just arrive for me try out), I have been told that the Mediatrix products have a working T38 implementation. Of course my suggestion would be check with the provider tho you plan to use the product with and see what they suggest/have seen work before. This is the base product... http://www.voipsupply.com/product_info.php?manufacturers_id=16products_id=334 I don't think the current Sipura firmware (for any model including the 2100) supports T38 yet. J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T.38 ATA
The newest 2100 firmware has T.38. What about other Sipura products like SPA-1001 and SPA-2002 ? Does it really have to be the one with broadband functionality integrated ? Thanks, Ivan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T.38 ATA
The MOSA 3700 family from Vodtel have working T.38. They come from 2 to 16 ports. Can be bought on www.bobascom.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moody Sent: den 14 september 2005 14:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T.38 ATA Can anyone recommend me ATA device that REALLY has T.38 built in. While I have not tested it myself (one just arrive for me try out), I have been told that the Mediatrix products have a working T38 implementation. Of course my suggestion would be check with the provider tho you plan to use the product with and see what they suggest/have seen work before. This is the base product... http://www.voipsupply.com/product_info.php?manufacturers_id=16products_id=334 I don't think the current Sipura firmware (for any model including the 2100) supports T38 yet. J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE Configuration problems
Kevin Bockman wrote: I'm having some problems getting TDMoE setup for the 1st time. I have a TE405P installed in the main server with an ethernet cross-connection to the secondary machine. (Yes, I know about IAX2 but I want to use TDMoE to simulate using T1s.) I'm using -HEAD from yesterday. On the main machine /etc/zaptel.conf: loadzone = us defaultzone=us dynamic=eth,eth1/00:30:48:84:74:25,24,0 bchan=1-23 dhcan=24 If you loaded wct4xxp before ztd-eth/ztdynamic your channels should be: 1-96 TE405P 97-120 TDMoE *CLI zap show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 Dynamic 'eth' span at 'eth1/00:30:48:84· RED 0 0 0 Leonardo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN vs NAT Helper
I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don't know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Application Return Codes - Help needed
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-CHANUNAVAIL,1,NoOp(CHANUNAVAIL)
exten = s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-CONGESTION,1,NoOp(CONGESTION)
exten = s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-ANSWER,1,NoOp(ANSWER)
exten = s-ANSWER,2,UserEvent(Answer|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-BUSY,1,NoOp(BUSY)
exten = s-BUSY,2,UserEvent(Busy|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-NOANSWER,1,NoOp(NOANSWER)
exten = s-NOANSWER,2,UserEvent(NoAnswer|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
Outbound calls are made using Manager originate interface from a meetme
room channel Local/4000/n where 4000 is an extension which accesses the
meetme room.
ITSP is terminating outbound calls to me via IAX2.
I need to be able to see the CAUSE CODE status of the call if it is answered, CONGESTED or BUSY.
my ITSP is in Australia - as am I.
the IAX2 debug clearly indicates a zero CAUSE CODE on most call cases.
Any idea what I might be able to do to make the CAUSE CODE a little more meaningful?
Cheers,
Mark.
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Re: [Asterisk-Users] Meetme Question
Hi, Thank you very much for your suggestion this was what i nedded. Best Regards Accursio Avona The question is, how can i indicate the marked user? A quick search of the archives reveals: Example: meetme.conf conf = 1000 extensions.conf ; ** Normal users enter the conference here ** exten = 4823,1,SetMusicOnHold(random) exten = 4823,2,Meetme(|Msciw) exten = 4823,3,Hangup() ; ** Extension to mark conference users* exten = 4824,1,Authenticate(12345) exten = 4824,2,Meetme(|Asci) exten = 4824,3,Hangup() Users using extension 4823 and entering conference 1000 will listen to hold music until the marked users enters. Users using extension 4824 and entering a password of 12345 will be able to select conference 1000 as the marked user. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI to PRI passthrough with DID intact
I currently have:Telco-PRI Panasonic DBS576 PBX EM wink T1 Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least expensive solution: Telco-PRI Asterisk PRI Panasonic DBS576 PBX. I have a second Digium T1/PRI card available. I am going to make the change after hours and know that I may have things to fiddle with after it is in use, but my biggest concern is getting my current Panasonic DIDs to come over from the Asterisk. I know that I can make a DID list pointing to the Panasonic extensions and assume that the dialplan will send the to the Panasonic. But I think this will be a dialed extension and not a DID call. But, I am assuming that there is another feature that can see the DID from the Telco and forward the call to the Panasonic as a DID call. I guess I am saying that I am not sure what the best option is. Please advise. PS. The Asterisk is still in testing phase and the people with Asterisk extensions know that they may not always have service, but when I make this change, the Panasonic MUST still be fully functional. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation)
I would be most interested in seeing some TNT/APX configurations and corrosponding SIP configurations for Asterisk. Right now, I'm using call routes and switching off a T1/PRI to my asterisk box, and would love to change that to pure SIP if possible. The only caveat is that my TNT boxes are primarily used for dialup traffic. Also, on the TNT, I see calling name information coming in from the PSTN (Lucent 5E), but the TNT will not pass it through the PRI to my * box. Am I understanding correctly that calling name information also does not work with SIP? Thanks, -- Troy Settle Pulaski Networks 866.477.5638 http://www.psknet.com Damon Estep wrote: If you are looking for real high density VOIP termination I would look at something like a Lucent APX 8000, configure correctly it can pass 2500+ g.729 calls to the PSTN course we paid lots of $ for ours. Chris Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is outbound it is fine, but what about origination? Are you using something other than asterisk as a SIP proxy? On a smaller scale the TNT is a good bet since the number of calls it will do (672 with t3) is closer to what an asterisk box can do without trans-coding. You can connect 1 partially populated TNT to one * box and not need another sip proxy, you can also have a failover sip proxy configured but not active unless the primary fails to respond. Both the TNT and APX have issues with calling name delivery over PRI when connected to a Lucent 5ESS configured to do end office LIDB dips, so calling party name on inbound calls can be a bear, look to connect to a Nortel DMS if you have the option -- go figure the LUCENT media gateways work better with Nortel class 5's than then they do with lucent class 5's. Have you learned something I have not about how to get all of the calls a TNT/APX can handle terminated on the SIP side without still having a single point of failure in the SIP proxy? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 ATA
I can confirm that sipura spa-2100 has t.38 suppurt from firmware 3.2.1 and it seems to work fine in our test with some t.38 providers. Bye Rosario - Original Message - From: Nenad Radosavljevic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, September 14, 2005 6:58 AM Subject: [Asterisk-Users] T.38 ATA Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that from Sipura release notes for firmwares. Anything else (other then Cisco routers with FXS modules) with T.38, or at least can someone confirm me that Sipura SPA-2100 has T.38 (firmware version would be nice info also) ? Thank you very much. Nenad Radosavljevic ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN vs NAT Helper
If you have a linux box, then u can try sip-nat-helper for netfilter. Cheers. Mensaje citado por: Waldo Rubinstein [EMAIL PROTECTED]: I\'m wondering if anyone can recommend one over the other. I\'m mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don\'t know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y participá de todos los beneficios del Portal Arnet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf Syntax to match first digits
ah, i see. didn't stumble over this yet, thanks! On Wed, 14 Sep 2005 13:48:37 +0200 [EMAIL PROTECTED] wrote: take a look into the wiki... http://www.voip-info.org/wiki-Asterisk+variables ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Callerid fails in any release after beta1 fails
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 14, 2005 4:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Callerid fails in any release after beta1 fails Richard Kashdan wrote: On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote: I am having the identical problem. I use the CVSHEAD Asterisk and do an update every couple of weeks or so. I did one last week and the caller id quit working on my two lines that have x100p cards. I didn't make any changes to my configuration files at that time, simply updated Asterisk. In the meantime I checked my configuration files carefully and don't see anything wrong. Callerid has stoped working for us as well from the SIP phones to the PRI. PRI to the SIP phones work fine. Doug Today I did a make update for zaptel, libpri and asterisk. Then recompiled. I no longer get an error message. Callerid is still blank. The log and cli return this line: Sep 14 08:22:51 NOTICE[13266]: chan_zap.c:5946 ss_thread: Got event 18 (Ring Begin)... I was getting a checksum error and a mylen 0 error. It would say callerid failed: success. I deleted all modules and did a make install of the beta1 source using the cvshead of zaptel and libpri. Caller id then works fine? Something has changed in the asterisk code that is not seeing callerid from of my x101p. I'm stumpted! --John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] first character in line 11 missing
On Monday 12 September 2005 01:56, Ronald Wiplinger wrote: I would like to know if somebody else experienced that: sip show peers will always drop the first character of the 11th line. while sip show peers like [0-9,a-z] will not drop any character. Can anybody test this, please? I have also noticed that the command database show also displays some lines without the first character (which should always be '/'). I am using 1.0.9/bristuffed 8l Paul bye Ronald Wiplinger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN vs NAT Helper
I think STUN is quite widely supported by hardphones. I'd be interested to know if STUN is a magic fix to SIP NAT - I've a feeling that its not. Derek Waldo Rubinstein wrote: I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don't know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 244 9719 United Kingdom: 0870 068 2368 International: 00 353 1 244 9719 Derek Conniffe DDI: 01 201 0146 (International: 00 353 1 201 0146) Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com begin:vcard fn:Derek Conniffe n:Conniffe;Derek org:Rivertower Ltd;IT adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 201 0146 tel;fax:+353 1 201 0085 tel;cell:+353 86 856 3823 note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A= Ireland: (Local) 01 244 9719=0D=0A= United Kingdom: 0870 068 2368=0D=0A= International: 00 353 1 244 9719=0D=0A= url:http://www.rivertowerhosting.com version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] with Eyebeam
On Wed, 14 Sep 2005, Dinesh wrote: I was wondering if its possible to hook up eyebeam with video support to [EMAIL PROTECTED] Yes. But eyebeam's video support is pretty rudimentary. it doesn't show inbound video at all until you start sending yours. people with eyebeam but no camera can't receive video at all. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first extensions (100). this is a asterisk problem? or a pbx
problem?
___
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Re: [Asterisk-Users] (no subject)
unless you show us some config files, I doubt that anybody can help you... On Wednesday 14 September 2005 16:46, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime IAX
On 9/2/05, Chris A. Icide [EMAIL PROTECTED] wrote: Dana Olson wrote: Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana Actually, the current CVS Head usage is rtupdate=yes|no, it was changed from rtnoupdate=yes|no not too long ago. If you are using 1.2 I'm not sure which is correct. I went through this battle of getting this to work the beginning of this week, and the four settings I listed in my last post made all the difference. -Chris Just to follow up with this thread, kpflemming provided the solution that I overlooked - the port column in the iax table was set to 0 instead of 4569. I didn't think to change it because the wiki said that the port, ipaddr, etc were all optional. For IAX peers, the port is not optional. I added a note to the wiki stating so as well. -- Dana ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation)
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Troy Settle Sent: Wednesday, September 14, 2005 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation) I would be most interested in seeing some TNT/APX configurations and corrosponding SIP configurations for Asterisk. www.voip-info.org - search for asterisk tnt Right now, I'm using call routes and switching off a T1/PRI to my asterisk box, and would love to change that to pure SIP if possible. The only caveat is that my TNT boxes are primarily used for dialup traffic. I have never tried a TNT for dual use, but it can be done. Might be too much CPU load if there are a lot of calls. Also, on the TNT, I see calling name information coming in from the PSTN (Lucent 5E), but the TNT will not pass it through the PRI to my * box. Am I understanding correctly that calling name information also does not work with SIP? Calling name does work with SIP. There is an issue with calling name delivery form a 5E to a TNT/APX if the 5E is configured to do end office LIDB dips for calling name (like qwest communications does it). The TNT does not understand the way that the 5E sends information following operation and subsequent facility IE containing the CNAM. BUT if you are seeing the CNAM on the TNT that may not be the issues, if this problem is present you usually will not see the name on the TNT either, just the number. Thanks, -- Troy Settle Pulaski Networks 866.477.5638 http://www.psknet.com Damon Estep wrote: If you are looking for real high density VOIP termination I would look at something like a Lucent APX 8000, configure correctly it can pass 2500+ g.729 calls to the PSTN course we paid lots of $ for ours. Chris Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is outbound it is fine, but what about origination? Are you using something other than asterisk as a SIP proxy? On a smaller scale the TNT is a good bet since the number of calls it will do (672 with t3) is closer to what an asterisk box can do without trans-coding. You can connect 1 partially populated TNT to one * box and not need another sip proxy, you can also have a failover sip proxy configured but not active unless the primary fails to respond. Both the TNT and APX have issues with calling name delivery over PRI when connected to a Lucent 5ESS configured to do end office LIDB dips, so calling party name on inbound calls can be a bear, look to connect to a Nortel DMS if you have the option -- go figure the LUCENT media gateways work better with Nortel class 5's than then they do with lucent class 5's. Have you learned something I have not about how to get all of the calls a TNT/APX can handle terminated on the SIP side without still having a single point of failure in the SIP proxy? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] timeout with queue
Hi, I've setup a queue with 3 sip members. I've tried with random and roundrobin and different timeout settings in musiconhold.conf Always after the second Nobody picked up in 15000ms I get Exiting on time-out cycle Stopped music on hold on CAPI/contr1/s-0 Where can I increase this timeout? asterisk 1.0.9 on linux 2.6.11 SuSE 9.3 Thanks a lot Regards Wolfgang ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards
como'n folks.. ... Well, as I told earlier.. my asterisk was running great with one fxo and one fxs module of a TDM400P All i tried last night to run asterisk with non-root I must did something wrong while I was trying to do that FXO module on channel # 1 FXS module on channel # 4 /etc/zaptel.conf - loadzone = us defaultzone=us fxoks=1 fxsks=4 /etc/asterisk/zapata.conf signalling=fxo_ks channel=1 signalling=fxs_ks channel=4 Did I made any mistake above? /etc/modprobe.conf - install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg # there have more line in it.. i guess they are not important here alias wcfxs wctdm /etc/rc.d/init.d/asterisk - #important lines are below start) /sbin/modprobe wctdm daemon /usr/sbin/asterisk stop) killproc asterisk /sbin/modprobe -r wctdm Here is the output of asterisk -vvvc Sep 13 22:18:11 WARNING[3982]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such device Sep 13 22:18:11 ERROR[3982]: chan_zap.c:6612 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Sep 13 22:18:11 ERROR[3982]: chan_zap.c:9990 setup_zap: Unable to register channel '1' Sep 13 22:18:11 WARNING[3982]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Sep 13 22:18:11 WARNING[3982]: loader.c:543 load_modules: Loading module chan_zap.so failed! see I have channel # 1 [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw-rw 1 root asterisk 196, 1 Sep 13 22:20 1 crw-rw 1 root asterisk 196, 2 Sep 13 22:20 2 crw-rw 1 root asterisk 196, 3 Sep 13 22:20 3 crw-rw 1 root asterisk 196, 4 Sep 13 22:20 4 crw-rw 1 root asterisk 196, 254 Sep 13 22:20 channel crw-rw 1 root asterisk 196, 0 Sep 13 22:20 ctl crw-rw 1 root asterisk 196, 255 Sep 13 22:20 pseudo crw-rw 1 root asterisk 196, 253 Sep 13 22:20 timer After all of this, if I comment out this from /etc/asterisk/zapata.conf /etc/asterisk/zapata.conf ;signalling=fxo_ks ;channel=1 signalling=fxs_ks channel=4 My asterisk run fine .. I just dont able to use my phone set attaced to my fxs module. And all I had to do is .. forward all incoming call to my voicemail box !! Should I consider my fxs card has burnt out !! Thanks for reading.. hope someone will reply me to help. Thanks again, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri release cause code mismatch
Hi ! Asterisk sends a RELASE COMPLETE with cause code 34. It seems that Nortel expects a RELEASE message in this state. The conversion is done in the protocol engine of the MSDL. Why would you want the cause code 34 to be sent ? Do you need a special rerouting on the Nortel side ? Would it be a help if you send a cause 3 ? (RELASE msg) Best regards Hans Tirpák Miklós schrieb: Hi! My asterisk (1.0.7) is connected to a Nortel pbx with Digium E100P card, both side are ETSI EuroISDN. I would like to reject an incomming call with cause code 34, but the Nortel PBX gets the value of 31 instead of 34. It seems to work on the asterisk side: Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 17162/0x430A) (Originator) Message type: SETUP (5) ... Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 49930/0xC30A) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 a2] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (2) ] My macro looks like: exten = s,1,SetVar(PRI_CAUSE=34) exten = s,2,Hangup According to the debug on Nortel it gets 31 cause code in the release complete q.931 message. Do you have any idea? Thanks, Miklos ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS using a PRI channel
Hi, I have some experience in sending SMSs using smsclient. I call the german Vodafone SMSC (01722278020), and smsclient takes approx 20 secs to send a SMS. The hardware is an Sedlbauer ISDN card. Now, I want to do the same using asterisk and a digium PRI card. I dialed using the manager with: action: originate channel: Zap/g4/01722278020 ... I assumed, the call will fail, because the remote end will become signalled a voice call, and imho the SMSC wouldn't answer a voice call, but expects data calls. Well, originating succeeded, and the respective context in the dialplan was accessed: -- Executing SMS(Zap/94-1, me||mycellnr|Test) in new stack -- Executing NoOp(Zap/94-1, Done) in new stack The application SMS returned without error, but returned immedeately (much less than 1 sec.). Of course, no SMS was sent. How can I debug this? How can I force Zap to data mode. The d option seem to be something different. Did anybody try sending SMS to german Vodafone or other SMSC mentioned in the smsclient package? Thanks for hints! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in
the pbx. and all incomming calls go to 100. thats the problem i will
try to solve this.
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first extensions (100). this is a asterisk problem? or a pbx
problem?
___
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---end quoted text---
--
.-
Pablo Allietti
LACNIC
___
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[Asterisk-Users] IAX Registration with servers
I have 2 servers that I use to talk from one place to another place. One of them, Server A registers with the other one, Server B. There are many cases the registration drops out and then works again after some time. The internet connection between them is not so great, which could be suspected. Server A also registers with VoicePulse. The connection to VoicePulse always works. It is only the connection with server B that fails often. Is there a timeout period that can be adjusted to maintain the connection despite bad internet? Is there a way to force the connection attempts to be more frequent? Here is the configuration on the 2 servers. iax.conf on server A ; Register with Indidge US server register = serverB:[EMAIL PROTECTED] [serverB] type=friend auth=md5 secret=password host=11.11.11.11 context=ctx disallow=all allow=ilbc qualify=yes notransfer=yes iax.conf on server B [serverA] type=friend auth=md5 secret=password host=dynamic context=ctx disallow=all allow=ilbc qualify=yes notransfer=yes If anyone could suggest some solution, it is appreciated. Thank you. Sincerely, -- Naren Koka VP of Technology INDIDGE SYSTEMS (480) 829-0479 x111 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Registration with servers
On Wed, 2005-09-14 at 08:10 -0700, Naren Koka wrote: I have 2 servers that I use to talk from one place to another place. One of them, Server A registers with the other one, Server B. There are many cases the registration drops out and then works again after some time. The internet connection between them is not so great, which could be suspected. Server A also registers with VoicePulse. The connection to VoicePulse always works. It is only the connection with server B that fails often. One important piece of info missing, are they on fixed IPs? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.9 long term stability
I've been evaluating asterisk for quite some time now and am attempting to create services on it. The system is simple right now. asterisk seems to look up atleast every week if not more. I am running asterisk 1.0.9 and would like to find similiar experiences of long term stability. I attempted to debug it, but my asterisk isn't compiled with all the possible debugging flags, which flags in the Makefile should I enable to help provide more information? Here is what I have found so far. gdb attach backtrace:(gdb) bt #0 0x401c4a76 in nanosleep () from /lib/libc.so.6 #1 0x000c in ?? () #2 0x401ef4ba in usleep () from /lib/libc.so.6 #3 0x in ?? () #4 0x8a9fa304 in ?? () #5 0x8a9ffbe0 in ?? () #6 0x in ?? () #7 0x03e8 in ?? () #8 0x8a9fa504 in ?? () #9 0x40678bbb in zt_handle_event (ast=0x8a9fa504) at chan_zap.c:590 Previous frame inner to this frame (corrupt stack?) (gdb) info threads This is definitely something in zaptel (zt_handle) but the other errors like corrupt stack lead me to believe there is also something else wrong. Any input would be greatly appreciated.-- Sig Langehttp://www.signuts.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid fails in any release after beta1 fails
John Hill wrote: I deleted all modules and did a make install of the beta1 source using the cvshead of zaptel and libpri. Caller id then works fine? Something has changed in the asterisk code that is not seeing callerid from of my x101p. I was thinking about doing a fresh install this weekend as well to see if that makes any difference. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] R1.502 of chan_zap.c kills callerid on a x101p
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 14, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Callerid fails in any release after beta1 fails John Hill wrote: I deleted all modules and did a make install of the beta1 source using the cvshead of zaptel and libpri. Caller id then works fine? Something has changed in the asterisk code that is not seeing callerid from of my x101p. I was thinking about doing a fresh install this weekend as well to see if that makes any difference. Doug R1.502 of chan_zap.c kills callerid on a x101p You might want to wait. I'm trying to figure out how to report this as a bug. --john ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P stops answering
Kevin P. Fleming wrote: Andy Howell wrote: I have a weird problem in which my digium card stops answering. After running for a couple days, incoming calls are not seen. Running asterisk -r shows no incoming calls. Restarting Asterisk does not help. After a reboot it is fine. This problem was fixed in CVS (HEAD and v1-0) quite some time ago; what versions are you running? Its 1.0.9, as part of [EMAIL PROTECTED] 1.3 Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P stops answering
Leonardo Gomes Figueira wrote: Hi, Andy Howell wrote: I have a weird problem in which my digium card stops answering. After running for a couple days, incoming calls are not seen. Running asterisk -r shows no incoming calls. Restarting Asterisk does not help. After a reboot it is fine. Any ideas? Do you have APIC enabled on the BIOS/kernel ? Try to disable it on the BIOS or with noapic on the kernel. I found out this was the cause of this problem here on a VIA motherboard and it was fixed with noapic. I just don't know why... :) Leonardo, I have it APIC disabled. I thought that interupts might be the problem from reading the voip wiki. I had the card in another machine, where I first noticed the problem. After lots of messing around, I decided to go with a machine that others said worked well. I'm now running on a Dell Optiplex GX150 with 1Ghz CPU and 512MB of memory. The machine is dedicated to asterisk. At boot, the card is reported as: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) I suppose I could just reboot nightly. Trouble is, there is no way to detect that it is not working, other than trying to call in. Outgoing calls continue to work. Thanks, Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards
On Tue, Sep 13, 2005 at 12:01:21PM -0800, Mojo with Horan Company, LLC wrote: hisax seems to be a loadable module for an ISDN card. if: # lsmod | grep hisax prints any output, try # rmmod hisax; modprobe zaptel What information does kudzu use? Why doesn't it know that those PCI IDs are used for zaptel as well? Doesn't it update its modules information from the installed modules? Also: isn't there a simple way to load some modules in andvance and/or blacklist others? (/etc/modules and /etc/hotplug/blacklist , respectively on debian). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P - [EMAIL PROTECTED] Install Problems
Title: TE110P - [EMAIL PROTECTED] Install Problems I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single [EMAIL PROTECTED] system with a single T1 card. Robbed Bit T1 ami, d4. --inbound call -- Starting simple switch on 'Zap/7-1' -- Starting simple switch on 'Zap/14-1' -- Executing Playback(Zap/7-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Macro(Zap/7-1, hangupcall) in new stack -- Executing ResetCDR(Zap/7-1, w) in new stack -- Executing NoCDR(Zap/7-1, ) in new stack -- Executing Wait(Zap/7-1, 5) in new stack -- Executing Playback(Zap/14-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Hangup(Zap/7-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/7-1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'Zap/7-1' -- Hungup 'Zap/7-1' -- Executing Macro(Zap/14-1, hangupcall) in new stack -- Executing ResetCDR(Zap/14-1, w) in new stack -- Executing NoCDR(Zap/14-1, ) in new stack -- Executing Wait(Zap/14-1, 5) in new stack -- Executing Hangup(Zap/14-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/14-1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' --outbound call -- Executing SetVar(SIP/4901-cd04, OUTNUM=mynum) in new stack -- Executing Cut(SIP/4901-cd04, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/4901-cd04, 0?19) in new stack -- Executing Dial(SIP/4901-cd04, ZAP/g0/mynum) in new stack -- Called g0/mynum -- Zap/1-1 answered SIP/4901-cd04 -- Hungup 'Zap/1-1' --- [zaptel.conf] span=1,1,0,d4,ami # have tried with 1,0,0 - same problem fxsks=1-24 loadzone = us defaultzone=us [zapata.conf] [channels] signalling=fxs_ks group=0 ;context=incoming channel=1-24 echocancelwhenbridged=yes echotraining=400 context=default faxdetect=incoming ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialplan Design Q
i guess is usefull a neighcompany context, where you will allow users
to call other companies, using a company prefix. I need more info about
your real dial patterns in order to suggest something more specific.
best regards
On 9/13/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have to design a dialplan for mulitple contexts (multiple companies)and I'm not sure how to go about it and I thought someone may offerhelp.Here is some background. There are three separate companies,let's say A, B and C.Each has their own context and each has their own
set of numbers (these are just examples, not the actual config):[ContextA]exten = 10,1,Dial(SIP/10,20)exten = 11,1,Dial(SIP/11,20)exten = 12,1,Dial(SIP/12,20)include = outbound
[ContextB]exten = 20,1,Dial(SIP/20,20)exten = 21,1,Dial(SIP/21,20)exten = 22,1,Dial(SIP/22,20)include = outbound[ContextC]exten = 30,1,Dial(SIP/30,20)exten = 31,1,Dial(SIP/31,20)
exten = 32,1,Dial(SIP/32,20)include = outbound[default]exten = _1X,1,GoTo(ContextA,${EXTEN},1)exten = _2X,1,GoTo(ContextA,${EXTEN},1)exten = _3X,1,GoTo(ContextA,${EXTEN},1)
[outbound]exten = _9XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])So each user registers and they can call each other and they can dial9xx to dial local and ld.The issue arises when they want/need
to call the other companies in the other contexts.I want the call togo direct to the other user instead of out our gateway and back in (likeit is happening now).I could go into each context and add the numbers
for the other users, but that doesn't scale very well.If I have 10different contexts and each has 4 phones, that's 40 entries per context.I am looking for a fairly easy way to do this.Any ideas?(note that
the extensions listed 10,11,20,30, etc are really 10 digits, I justdidn't want to have to type them all out).PA___--Bandwidth and Colocation sponsored by
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Re: [Asterisk-Users] TDM400P stops answering
Andy Howell wrote: Its 1.0.9, as part of [EMAIL PROTECTED] 1.3 Then I would suggest upgrading to 1.0.9.1 or the just-released 1.0.9.2. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to create IVR menu and transfer to another sip extensions.
mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to.On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote: Hi All, I need help to create one IVR Menu, when a say Welcome to PBX Corp... , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 ATA
- Original Message - From: Rosario Pingaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 14, 2005 9:07 PM Subject: Re: [Asterisk-Users] T.38 ATA I can confirm that sipura spa-2100 has t.38 suppurt from firmware 3.2.1 and it seems to work fine in our test with some t.38 providers. Are they pay-as-you-go providers? If so, do you mind sharing their names with us? Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.9 long term stability
Well I don't know how you could measure long term stability at the moment since 1.0.9 has only been out for about 2 months, but I can offer some insight on older versions. We have one 1.0.3 box that has been up and running for 27+ weeks without an issue. It is running SIP for ~ 30 phones and 1 Cisco gateway. [EMAIL PROTECTED] asterisk -rx show uptime System uptime: 27 weeks, 1 day, 13 hours, 52 minutes, 57 seconds Last reload: 3 weeks, 5 days, 18 hours, 29 minutes, 4 seconds [EMAIL PROTECTED] # asterisk -rx show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i686 running Linux We have another box that is running 1.0.7 with H.323 to an H.323 gatekeeper and it is just acting as voicemail for a Cisco Call Manager. It crashes at least 1-2 times per week. Starting asterisk again brings it back up. I don't know why it happens and I have been unable to get anything useful from the logs. It just dies. That said, from what I've seen in the past, if you are running SIP, it is very stable. I know I've seen people mention that they have to restart it every week or so, but I haven't seen that so far. Peder Sig Lange wrote: I've been evaluating asterisk for quite some time now and am attempting to create services on it. The system is simple right now. asterisk seems to look up atleast every week if not more. I am running asterisk 1.0.9 and would like to find similiar experiences of long term stability. I attempted to debug it, but my asterisk isn't compiled with all the possible debugging flags, which flags in the Makefile should I enable to help provide more information? Here is what I have found so far. gdb attach backtrace: (gdb) bt #0 0x401c4a76 in nanosleep () from /lib/libc.so.6 #1 0x000c in ?? () #2 0x401ef4ba in usleep () from /lib/libc.so.6 #3 0x in ?? () #4 0x8a9fa304 in ?? () #5 0x8a9ffbe0 in ?? () #6 0x in ?? () #7 0x03e8 in ?? () #8 0x8a9fa504 in ?? () #9 0x40678bbb in zt_handle_event (ast=0x8a9fa504) at chan_zap.c:590 Previous frame inner to this frame (corrupt stack?) (gdb) info threads This is definitely something in zaptel (zt_handle) but the other errors like corrupt stack lead me to believe there is also something else wrong. Any input would be greatly appreciated. -- Sig Lange http://www.signuts.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to create IVR menu and transfer to another sip extensions.
I need create one configuration to provide one Interactive Voice Response. I read any docs about this. So, if you have one sample, please post. Thanks. Paulo Santos. Brasil-RJMoises Silva [EMAIL PROTECTED] escreveu: mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote: Hi All, I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org " ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe!___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: T.38 ATA
Hi ! First of all thank you all for fast response on matter of T.38 capable ATAs. I have asked a UK VoIP suplier to check with manufacterers of various ATAs they sell, do they support T.38 and here is what they/I have got as a result: 1. Sipura SPA-2100 only and with firmware 3.2.1 is T.38 capable (no information on type of T.28 support UDPTL/TPKT) 2. All Gradnstream Handytone ATAs with firmware grater than 1.0.6.x are T.38 capable and they use UDPTL T.38 Regards, Nenad The newest 2100 firmware has T.38. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] actionID on manager events
Hello, all! I'm looking at the wiki page and info on the mailing list and I'm getting conflicting info... I am using the manager API from the telnet CLI and I am testing creating calls with it. I login with events: on and I can originate calls just fine. However, when I set ActionID on an Originate, I cannot see anywhere where that actionid carries into the Event output. But I found this on a post from January: Yes, ActionID is a value you can use when issuing a command. It there so that you can be sure you respond to your own responses not to someone else's or that you respond to an response instance in the correct way. In a multi-threaded app you might have several actions outstanding so you will need to know what response corresponds to which command. Which indicates that the actionid should be coming through. Is there perhaps some setting I'm missing? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.9 long term stability --thread hijack, why not reboot?
Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of After X days, Y thing does not work but if I reload or reboot, it's OK - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for me. We also do the Sunday reboot of all of our Windows servers as well as restarting all of the critical services such as IIS , SQL, Exchange etc nightly. It helps, a lot (Exchange is a notorious memory leaker) Of course, if your install processes calls 24/7 that's a different story. However, I expect that the majority of Asterisk installs are for a 9-to-5 type of operation. We run two shifts here, and we stop processing calls at 10 PM, and start again at about 6 AM - a large window of opportunity to reboot. Why not take advantage of it? I've also heard it said, something along the lines of: If you have to reboot, your server isn't set up correctly to which I say piffle. Even NASA has rebooted the Mars probes after they land and I understand that they run VXWorks, incidentally, the same RTOS that my Mitel 3300 uses, and *even Mitel* recommends periodic reboots, which we duly cron every night, 2 AM. 24/7/365 installs aside, is there a reason why reboots seem to be frowned upon? Again, not trolling, just curious. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.
This is a sample that I built as part of our * pilot here - it demonstrates the various things you can do with an auto-attendant type of system. Is this the kind of thing you are looking for? [info-line] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(demo-enterkeywords) exten = 1,1,Goto(library-info,s,1) exten = 2,1,Goto(lawn-sprinkling-info,s,1) exten = 3,1,Goto(closed-trails-info,s,1) exten = 4,1,Voicemail([EMAIL PROTECTED]) [library-info] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(demo-enterkeywords) exten = 1,1,Voicemail([EMAIL PROTECTED]) exten = 2,1,Goto(internal,96045551212,1) exten = 3,1,Playback(demo-congrats) exten = *,1,Goto(library-info,s,5) [lawn-sprinkling-info] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(demo-enterkeywords) exten = 1,1,Goto(internal,2348,1) exten = 2,1,Goto(closed-trails-info,s,1) exten = 3,1,Voicemail([EMAIL PROTECTED]) exten = *,1,Goto(lawn-sprinkling-info,s,5) [closed-trails-info] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Playback(demo-congrats) exten = s,6,Goto(info-line,s,5) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Sep 14, 2005, at 9:15 AM, PJ Santos wrote: I need create one configuration to provide one Interactive Voice Response. I read any docs about this. So, if you have one sample, please post. Thanks. Paulo Santos. Brasil-RJ Moises Silva [EMAIL PROTECTED] escreveu: mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote: Hi All, I need help to create one IVR Menu, when a say Welcome to PBX Corp... , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe!___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stupid tricks: preventable?
I just experienced something I'd rather not experience again. Using a SPA-841 SIP phone connected to our Asterisk server, someone dialed their own extension, answered, and then transferred the call using the phone's XFER soft key. This does a SIP REFER. Now, the phone has dropped out of the loop, and Asterisk has connected the two call legs into a loop, as far as I can tell. I tried a soft hangup on each of the channels, but nothing happened. Is there any way to recover from this, short of an asterisk restart? Thanks, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LiveVOIP - I win :)
LOL - Congrats! $30 down... Let's see... how much to go? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Monday, September 12, 2005 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] LiveVOIP - I win :) A few months ago, the friendly folks from liveVOIP went under. We had some discussion on how to limit our losses, and my recommendation was a chargeback, since FTTP Services -- their CC merchant -- wasn't affected by the bankruptcy, as far as we could tell. Today, I received this from my CC company: http://muware.com/asterisk/livevoip.pdf Anyone else got lucky? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!
I would recommend it for tech-savvy people right now. It's a bit klunky in the interface, but the phone functions (dial, receive, cid) work great and the sound is clear in both directions. the setup through the phone interface is a little repetitive and slow (albeit probably great for avid thumb-typers used to cellphones these days), and the web interface didn't have enough functions to allow complete setup. To give a feel for what I mean by 'klunky', it's just poor programming in the firmware. Change the ringer melody, volume, or style (ring or vibrate) and expect the phone to reboot all the way to use the new settings. I don't think there are actually _any_ settings you can change on the phone that _don't_ reboot it completely before taking effect. Oh yeah, this is the V2 model, came with (what, globug? gloworm? glopoint?) some proprietary firmware installed that wouldn't allow you to use your own sip server. I went to the manufacturer's web site, zyxel.com, and had no problems finding a firmware that alleviated the propriety. I believe it was the wj0011 that Rod mentions below; they seem to have a wv0001 now. Not sure if Rod had a V2 or the original model. Mojo Paul Hales wrote: I suppose the question is now whether you would recommend buying one later, PaulH - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 13, 2005 5:22 PM Subject: Re: [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD! An update on this... I was wrong. The wireless problem was an altogether different issue. the wj0011 firmware finally made my phone useable, after 6 months of problems. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax problems.
Hello. Im trying to get Fax-to-email working. I've installed Rx and txfax, spanDSP and every package needed. I've done everything on this page (altough, some bash-scripting problems): http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email anyway, when i try to send an fax, i get theese messages in asterisk: -- Executing Goto(SIP/5060-08148520, fax|2201|1) in new stack -- Goto (fax,2201,1) -- Executing Macro(SIP/5060-08148520, faxreceive) in new stack -- Executing SetVar(SIP/5060-08148520, FAXFILE=/var/spool/asterisk/fax/1126714845.5.tif) in new stack -- Executing DBget(SIP/5060-08148520, EXTEMAIL=2201/xEmail) in new stack -- DBget: varname=EXTEMAIL, family=2201, key=xEmail -- DBget: Value not found in database. -- Executing SetVar(SIP/5060-08148520, [EMAIL PROTECTED]) in new stack -- Executing Goto(SIP/5060-08148520, 7) in new stack -- Goto (macro-faxreceive,s,7) -- Executing RxFAX(SIP/5060-08148520, /var/spool/asterisk/fax/1126714845.5.tif) in new stack -- Executing System(SIP/5060-08148520, /var/lib/asterisk/scripts/mailfax *** *** FAX [EMAIL PROTECTED] /var/spool/asterisk/fax/1126714845.5.tif company) in new stack (edited out some personal details) When i look in /var/spool/asterisk/fax/. No *.tif file is created. The person sending faxes get's an error or alert on the fax machine. What can I do? I've chmodded (-R) /var/spool/asterisk/fax to 777. I'm trying to recieve Fax over the internet. I'm using gentoo. When I call the fax I hear something that sounds like a Modem 56k(--) dialing on the web. Please help me. Regards, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway?
How is this insecure? Most large business and wholesale providers use only IP authentication, relying on a session border controller to do the authentication work resulting in great scalability on the softswitch (since it does not have to act as a proxy as well). If they know your IP, and you know their IP, the only risk is that your IP address can somehow be hijacked. IP authentication is actually better when done with a SBC or firewall because it hides the SIP registration port from the hackers in the less than honest parts of the country/world. I do not think host= in asterisk has the same affect. It still listens and responds on 5060. If they do not know its there they cant try to hack it. I do agree that BOTH digest and IP authentication would be nice, but that is not the reality these days, my providers trust my IPs an I trust theirs, no need for auth as long as the routers in between remain secure. If someone hijacks my routes or theirs it is only a matter of seconds before we know it. If someone hijacks my auth credentials it may be a billing cycle or 2 before I figure it out. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, September 14, 2005 12:50 AM To: C. Savinovich; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway? What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static IP that these calls will be coming from and only accept calls from that IP, but that's still not quite as secure as digest authentication that would be available via registration. If you know what IP the calls are coming from, you simply insert a host=XX.XX.XX.XX instead of host=dynamic in your sip.conf for that peer and calls should then come in as they did before without them having to register. If they are pre-pending digits on to the front of what you're interpreting as the dialed number/extension, you may choose to lop them off in extensions.conf, but aside from that this is fairly straight forward. On 9/14/05, C. Savinovich [EMAIL PROTECTED] wrote: Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP.Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation.Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox SIP way which is: client registers to my proxy. Guess what, I can't find any samples on this!!, Can anyone please help?, I will probably need a sample sip.conf. and then, to make a test call, I can use another asterisk box and try asterisk to asterisk sip calls (without register) via the cli prompt. But I have no idea and I am intrigued. Thanks CS ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.9 long term stability
We have Asterisk 1.0 (CVS-v1-0-12/28/04-03:08:11 built by [EMAIL PROTECTED] on a i686 running Linux) and as a safe countermeasure we do a cron reboot every week. On four different locations. No more crashes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, September 14, 2005 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.0.9 long term stability Well I don't know how you could measure long term stability at the moment since 1.0.9 has only been out for about 2 months, but I can offer some insight on older versions. We have one 1.0.3 box that has been up and running for 27+ weeks without an issue. It is running SIP for ~ 30 phones and 1 Cisco gateway. [EMAIL PROTECTED] asterisk -rx show uptime System uptime: 27 weeks, 1 day, 13 hours, 52 minutes, 57 seconds Last reload: 3 weeks, 5 days, 18 hours, 29 minutes, 4 seconds [EMAIL PROTECTED] # asterisk -rx show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i686 running Linux We have another box that is running 1.0.7 with H.323 to an H.323 gatekeeper and it is just acting as voicemail for a Cisco Call Manager. It crashes at least 1-2 times per week. Starting asterisk again brings it back up. I don't know why it happens and I have been unable to get anything useful from the logs. It just dies. That said, from what I've seen in the past, if you are running SIP, it is very stable. I know I've seen people mention that they have to restart it every week or so, but I haven't seen that so far. Peder Sig Lange wrote: I've been evaluating asterisk for quite some time now and am attempting to create services on it. The system is simple right now. asterisk seems to look up atleast every week if not more. I am running asterisk 1.0.9 and would like to find similiar experiences of long term stability. I attempted to debug it, but my asterisk isn't compiled with all the possible debugging flags, which flags in the Makefile should I enable to help provide more information? Here is what I have found so far. gdb attach backtrace: (gdb) bt #0 0x401c4a76 in nanosleep () from /lib/libc.so.6 #1 0x000c in ?? () #2 0x401ef4ba in usleep () from /lib/libc.so.6 #3 0x in ?? () #4 0x8a9fa304 in ?? () #5 0x8a9ffbe0 in ?? () #6 0x in ?? () #7 0x03e8 in ?? () #8 0x8a9fa504 in ?? () #9 0x40678bbb in zt_handle_event (ast=0x8a9fa504) at chan_zap.c:590 Previous frame inner to this frame (corrupt stack?) (gdb) info threads This is definitely something in zaptel (zt_handle) but the other errors like corrupt stack lead me to believe there is also something else wrong. Any input would be greatly appreciated. -- Sig Lange http://www.signuts.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom randomly fails outbound calls,
Hi All, I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301 The Polycom misses 1 out of 2 dialout calls, this is the full log from a call which didn't go through. 303091 Sep 14 10:45:15 VERBOSE[15427]: -- SIP/pstn_2-1f35 answered SIP/200-0db1 303092 Sep 14 10:45:15 VERBOSE[15427]: -- Attempting native bridge of SIP/200-0db1 and SIP/pstn_2-1f35 303093 Sep 14 10:45:15 DEBUG[15427]: Ooh, format changed from unknown to ulaw 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found 303095 Sep 14 10:45:15 DEBUG[15427]: Ooh, format changed from unknown to ulaw 303096 Sep 14 10:45:15 DEBUG[15427]: Didn't get a frame from channel: SIP/pstn_2-1f35 303097 Sep 14 10:45:15 DEBUG[15427]: Bridge stops bridging channels SIP/200-0db1 and SIP/pstn_2-1f35 303098 Sep 14 10:45:15 DEBUG[15427]: update_user_counter(ww4902758) - decrement outUse counter 303099 Sep 14 10:45:15 DEBUG[15427]: ww4902758 is not a local user 303100 Sep 14 10:45:15 DEBUG[15427]: Exiting with DIALSTATUS=ANSWER. 303101 Sep 14 10:45:15 VERBOSE[15427]: == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on 'SIP/200-0db1' in macro 'dialout-trunk' 303102 Sep 14 10:45:15 VERBOSE[15427]: == Spawn extension (from-internal, 4902758, 1) exited non-zero on 'SIP/200-0db1' The Poly dials out using the SPA3000 FXO, all other phones connect to SPA300 FXO from SPA2000 FXS and they work fine when dialing out, What I noticed is that in the successful calls you could hear the tones going out, in the calls that fail there's only silence. I added two ww to check if it was a timing issue before getting tones, but is not. I guess the line 303096 is the more relevant, but I don't know where to start troubleshooting it. Any clue or tip will be appreciated, Thank you, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.
Posso ajudar? Fábio Sakai DGX - Digital Express Suporte CosmoCall [EMAIL PROTECTED] +55 11 3049.8109 De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de PJ Santos Enviada em: quarta-feira, 14 de setembro de 2005 13:16 Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions. I need create one configuration to provide one Interactive Voice Response. I read any docs about this. So, if you have one sample, please post. Thanks. Paulo Santos. Brasil-RJ Moises Silva [EMAIL PROTECTED] escreveu: mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote: Hi All, I need help to create one IVR Menu, when a say Welcome to PBX Corp... , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.0.9 long term stability
We have another box that is running 1.0.7 with H.323 to an H.323 gatekeeper and it is just acting as voicemail for a Cisco Call Manager. It crashes at least 1-2 times per week. Starting asterisk again brings it back up. I don't know why it happens and I have been unable to get anything useful from the logs. It just dies. That said, from what I've seen in the past, if you are running SIP, it is very stable. I know I've seen people mention that they have to restart it every week or so, but I haven't seen that so far. It's not necessarily any specific component of asterisk or version of asterisk that can cause instability. It might have nothing to do with asterisk. One of our boxes is running 1.0.7 (with SIP phones) and it has run like a champ for months with no incidents. I had another box that crashed frequently until I was able to come up with a good combination of hardware and software versions. There are so many factors involved. I read a post on this list not too long ago that said the poster has about 30 different asterisk boxes running, and they are all identical - asterisk version, asterisk patches, OS version, OS patches, telephony hardware, general computer hardware, right on down to the bios revision on the motherboard. That was the only way he was able to maintain consistent stability across all 30 boxes. I wish I had time to be that organized! - Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.
Thisisverybasicprogrammingandisexplainedinatutorials is you have a sip phone you will have a transfer or flash button so you can transfer any call to another a ivr menu is very simple to exten = s,1,Answer exten = s,2,Background(audiofile..) exten = 1,1,Dial(sip/100) exten =2,1,Dial(sip/101) press 1 to dial extension 100 press 2 to dial extension 101 it's as easy as thatand if you want an example you have to provide more details http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.htmlread this link and you will know all the basics Good luck Sander Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens PJ SantosVerzonden: woensdag 14 september 2005 18:16Aan: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionOnderwerp: Re: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions. I need create one configuration to provide one Interactive Voice Response. I read any docs about this. So, if you have one sample, please post. Thanks. Paulo Santos. Brasil-RJMoises Silva [EMAIL PROTECTED] escreveu: mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote: Hi All, I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org " ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax problems
Hello. Im trying to get Fax-to-email working. I've installed Rx and txfax, spanDSP and every package needed. I've done everything on this page (altough, some bash-scripting problems): http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email anyway, when i try to send an fax, i get theese messages in asterisk: -- Executing Goto(SIP/5060-08148520, fax|2201|1) in new stack -- Goto (fax,2201,1) -- Executing Macro(SIP/5060-08148520, faxreceive) in new stack -- Executing SetVar(SIP/5060-08148520, FAXFILE=/var/spool/asterisk/fax/1126714845.5.tif) in new stack -- Executing DBget(SIP/5060-08148520, EXTEMAIL=2201/xEmail) in new stack -- DBget: varname=EXTEMAIL, family=2201, key=xEmail -- DBget: Value not found in database. -- Executing SetVar(SIP/5060-08148520, [EMAIL PROTECTED]) in new stack -- Executing Goto(SIP/5060-08148520, 7) in new stack -- Goto (macro-faxreceive,s,7) -- Executing RxFAX(SIP/5060-08148520, /var/spool/asterisk/fax/1126714845.5.tif) in new stack -- Executing System(SIP/5060-08148520, /var/lib/asterisk/scripts/mailfax *** *** FAX [EMAIL PROTECTED] /var/spool/asterisk/fax/1126714845.5.tif company) in new stack (edited out some personal details) When i look in /var/spool/asterisk/fax/. No *.tif file is created. The person sending faxes get's an error or alert on the fax machine. What can I do? I've chmodded (-R) /var/spool/asterisk/fax to 777. I'm trying to recieve Fax over the internet. I'm using gentoo. When I call the fax I hear something that sounds like a Modem 56k(--) dialing on the web. Please help me. Regards, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] timeout with queue
In queues.conf ; How long do we let the phone ring before we consider this a timeout... ; timeout = 15 But this is just the function how long the phones will ring you should not set this option to long or your phone will stop ringing if a timeout is set in your phone But when the line hangs up after timeout you have set an option at the queue like this below it will stay in queue for 15 seconds then hangs up exten = 121,2,Queue(121|tT|||15) Exten = 121,3,Hangup -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Wolfgang Lumpp Verzonden: woensdag 14 september 2005 16:25 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] timeout with queue Hi, I've setup a queue with 3 sip members. I've tried with random and roundrobin and different timeout settings in musiconhold.conf Always after the second Nobody picked up in 15000ms I get Exiting on time-out cycle Stopped music on hold on CAPI/contr1/s-0 Where can I increase this timeout? asterisk 1.0.9 on linux 2.6.11 SuSE 9.3 Thanks a lot Regards Wolfgang ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: passing variables to h extension
Tony Mountifield ha scritto:
It works for me (using CVS HEAD, but I'm sure it's worked in the past for
me on Stable too). I think there must be some other reason it's not working
for you.
Just done a little test for it, as follows...
My extensions.conf:
[vartest]
exten = _X.,1,SetVar(FRED=hello)
exten = _X.,2,NoOp(FRED=${FRED})
exten = _X.,3,Playback(demo-congrats)
exten = _X.,4,Hangup
exten = h,1,NoOp(FRED=${FRED})
Yes it always worked also for me, using 1.2-beta1, typing error in noops
used for debug was having me look in the wrong place to set the vars !
Sorry for the rtfm question then
anyway I now wonder even more why accounting is done via cron jobs in
php-agi apps you find around isn't that only a waste of resources,
since you have to tag in some way calls already accounted ?
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RE: [Asterisk-Users] Stupid tricks: preventable?
i think you need a restart, then:
[your-local-extension-context]
exten = _,1,Gotoif([${CALLERIDNUM}=${EXTEN}]?2:4)
exten = _,2,Playback(you-are-a-frigging-idiot-stop-that)
exten = _,3,System(/etc/asterisk/email-administrator-moronic-behavior
${CALLERIDNUM})
exten = _,4,InsertNormalDialingBehaviorHere
I can't think of a reason why someone would want to dial their own extension
from their own extension, let alone transfer it, unless they want to leave
themselves voicemail??
Was this guy just trying to be a smart-alek?
-Original Message-
From: alan [mailto:[EMAIL PROTECTED]
Sent: Monday, September 12, 2005 3:22 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Stupid tricks: preventable?
I just experienced something I'd rather not experience again.
Using a SPA-841 SIP phone connected to our Asterisk server, someone
dialed their own extension, answered, and then transferred the call
using the phone's XFER soft key. This does a SIP REFER.
Now, the phone has dropped out of the loop, and Asterisk has connected
the two call legs into a loop, as far as I can tell.
I tried a soft hangup on each of the channels, but nothing happened.
Is there any way to recover from this, short of an asterisk restart?
Thanks,
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Re: T.38 ATA
about spa-2100, the t38 stream is on UDPTL and so asterisk passthrough doesn't work. - Original Message - From: Nenad Radosavljevic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, September 14, 2005 12:16 PM Subject: [Asterisk-Users] Re: T.38 ATA Hi ! First of all thank you all for fast response on matter of T.38 capable ATAs. I have asked a UK VoIP suplier to check with manufacterers of various ATAs they sell, do they support T.38 and here is what they/I have got as a result: 1. Sipura SPA-2100 only and with firmware 3.2.1 is T.38 capable (no information on type of T.28 support UDPTL/TPKT) 2. All Gradnstream Handytone ATAs with firmware grater than 1.0.6.x are T.38 capable and they use UDPTL T.38 Regards, Nenad The newest 2100 firmware has T.38. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom randomly fails outbound calls,
Hi Andres - I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301 The Polycom misses 1 out of 2 dialout calls, this is the full log from a call which didn't go through. 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found 303095 Sep 14 10:45:15 DEBUG[15427]: Ooh, format changed from unknown to ulaw 303096 Sep 14 10:45:15 DEBUG[15427]: Didn't get a frame from channel: SIP/pstn_2-1f35 303097 Sep 14 10:45:15 DEBUG[15427]: Bridge stops bridging channels SIP/200-0db1 and SIP/pstn_2-1f35 I guess the line 303096 is the more relevant, but I don't know where to start troubleshooting it. Line 303095 is probably relevant, too. What codec is the phone configured to try first? It looks like the phone is trying to use something asterisk doesn't understand, or is not configured for. Maybe set the phone to ulaw instead. Also, what dtmfmode are you using? Can we look at your sip.conf from asterisk, and the config files for your Polycom phone? - Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax problems
On Wed, 2005-09-14 at 19:15 +0200, Arne Morten Johansen wrote: Hello. Im trying to get Fax-to-email working. Didn't I see that exact same message exactly 29 minutes ago? That's the best way _not_ to get an aswer on this list. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk 1.0.9 long term stability
lol that was me ironic that I just hijacked this thread and said that reboots are not a bad thing! It's true I do have 30 IAX/SIP boxen that I don't reboot, they are all slave servers to the IAX/SIP/PRI master server, which I *do* reboot every night. The 30 boxen I did by cloning a single hdd and making specific changes to each box. this underscores the importance of documenting everything, which I did for the 30, after I had what I felt was a good config, I documented each single step to clone the config, getting right down to things like reflashing the bios, and making sure that a specific brand and rev of nic was in a specific slot, and that nic used a specific IRQ, and that certain hdw like the onboard sound card in the chassis was disabled. Armed with such a document, a third party can come in and easily re-create your work, or troubleshoot it. The trick is getting the config just right in the first place. As I have said before, a lot of these issues can be resolved before they happen simply by not using random hardware and random distro, and to that end, it might serve Digium well to have a posted HCL and / or sell a certified barebones box and certify Asterisk on a specific RedHat and / or Debian, and if you use anything else, well, tough. I, myself, would prefer to buy this cerified solution instead of sweating through a from-scratch config and crossing your fingers and hoping it is stable. I'm sure a lot of other guys on this list would jump on it as well. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 14, 2005 11:00 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk 1.0.9 long term stability We have another box that is running 1.0.7 with H.323 to an H.323 gatekeeper and it is just acting as voicemail for a Cisco Call Manager. It crashes at least 1-2 times per week. Starting asterisk again brings it back up. I don't know why it happens and I have been unable to get anything useful from the logs. It just dies. That said, from what I've seen in the past, if you are running SIP, it is very stable. I know I've seen people mention that they have to restart it every week or so, but I haven't seen that so far. It's not necessarily any specific component of asterisk or version of asterisk that can cause instability. It might have nothing to do with asterisk. One of our boxes is running 1.0.7 (with SIP phones) and it has run like a champ for months with no incidents. I had another box that crashed frequently until I was able to come up with a good combination of hardware and software versions. There are so many factors involved. I read a post on this list not too long ago that said the poster has about 30 different asterisk boxes running, and they are all identical - asterisk version, asterisk patches, OS version, OS patches, telephony hardware, general computer hardware, right on down to the bios revision on the motherboard. That was the only way he was able to maintain consistent stability across all 30 boxes. I wish I had time to be that organized! - Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: (no subject)
This is not a siemens pbx problem you set the
pridialplan = to national and that adds a number to the outgoing call or
something just use
Pridialplan = local
prilocaldialplan = local
and it should work
I tried to open the file kds again and now it showed me your configuration
:) don't know why it did not show me before
Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 17:31
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the
pbx. and all incomming calls go to 100. thats the problem i will try to
solve this.
It could potentially be both. I would look at your extensions.conf
first though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example
100
102 etc. but when i truy to go outside with the 9 before the call
rings in the first extensions (100). this is a asterisk problem? or
a pbx problem?
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LACNIC
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[Asterisk-Users] Re: TE110P - [EMAIL PROTECTED] Install Problems
On Wed, Sep 14, 2005 at 10:45:36AM -0500, Robert Wagner wrote: hi te110p is a et1 card. your sigfnalling is wrong i think i have the same card and is work with this conf /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] context=from-pstn rxwink=400 ; Atlas seems to use long (250ms) winks relaxdtmf=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;group=1 ;callgroup=1 ;pickupgroup=1 signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 switchtype=national echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 ;context=default ; Points to the default context of your extensions.conf channel = 1-15,17-31 ; Set this to 1-15,17-31 for E1 I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single [EMAIL PROTECTED] system with a single T1 card. Robbed Bit T1 ami, d4. --inbound call -- Starting simple switch on 'Zap/7-1' -- Starting simple switch on 'Zap/14-1' -- Executing Playback(Zap/7-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Macro(Zap/7-1, hangupcall) in new stack -- Executing ResetCDR(Zap/7-1, w) in new stack -- Executing NoCDR(Zap/7-1, ) in new stack -- Executing Wait(Zap/7-1, 5) in new stack -- Executing Playback(Zap/14-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Hangup(Zap/7-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/7-1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'Zap/7-1' -- Hungup 'Zap/7-1' -- Executing Macro(Zap/14-1, hangupcall) in new stack -- Executing ResetCDR(Zap/14-1, w) in new stack -- Executing NoCDR(Zap/14-1, ) in new stack -- Executing Wait(Zap/14-1, 5) in new stack -- Executing Hangup(Zap/14-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/14-1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' --outbound call -- Executing SetVar(SIP/4901-cd04, OUTNUM=mynum) in new stack -- Executing Cut(SIP/4901-cd04, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/4901-cd04, 0?19) in new stack -- Executing Dial(SIP/4901-cd04, ZAP/g0/mynum) in new stack -- Called g0/mynum -- Zap/1-1 answered SIP/4901-cd04 -- Hungup 'Zap/1-1' --- [zaptel.conf] span=1,1,0,d4,ami # have tried with 1,0,0 - same problem fxsks=1-24 loadzone = us defaultzone=us [zapata.conf] [channels] signalling=fxs_ks group=0 ;context=incoming channel=1-24 echocancelwhenbridged=yes echotraining=400 context=default faxdetect=incoming ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Indications for Ireland
Hello asterisk-users, Just curious if anyone has the indications for Ireland, tried googling for it to no avail. Sean -- +---+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---+ Strange things happen under the midnight sun when Men and Dogs go hunting for gold smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Consulting Project ISO Hired Gun
Have a customer with a fairly large scale project that needs to get done, yesterday. Not sure how they thought they would be able to complete this internally, but they have basically a week or so to pull this off. Here is a list of requirements, if someone is interested in taking this on, preference would be for a single individual or firm to handle the job. Specifics are available, here is the overview. 1. We need an Engineer On-Call for a week for installation assistance (Asterisk Related) 2. Would like the Network Topology and Segmenting Double checked for accuracy and reliability. 3. Ensure proper connectivity with a Brook Trout Fax Server 4. Ensure interconnectivity with Modem Pool 5. Custom AGI script authoring MSSQL /mySQL to store call data 6. Custom script Training - What was done custom overview of why! 7. Setup of 3 T1 by working with CO 8. Custom Recording Application - files exported to NAS server w/naming convention to look-up calls i.e.800#,agentID,date,time 9. Someone available to work with us to ensure a smooth roll out. If anyone here feels they could handle this project, and can provide a few references please email me. Regards, -- Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax/TxFax - Compile Problem
Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.0.9 long term stability
In article [EMAIL PROTECTED], [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We have another box that is running 1.0.7 with H.323 to an H.323 gatekeeper and it is just acting as voicemail for a Cisco Call Manager. It crashes at least 1-2 times per week. Starting asterisk again brings it back up. I don't know why it happens and I have been unable to get anything useful from the logs. It just dies. Which H.323 stack are you using? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sox conversion has introduces background hiss for both 8k and 41K recordings to gsm
I took a recording that was in 41k sampled mono wav. Did the sox file.wav -r 8000 file.gsm resample ql took an 8K record in wave did sox file2.wav file2.gsm Both of them have introduced a hissing noise. If I play the wave files they sound fine. How do I remove or reduce the hiss. Jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Consulting Project ISO Hired Gun
I am game. What do you need from me??? Locked, loaded and ready to GO!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Wednesday, September 14, 2005 2:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Consulting Project ISO Hired Gun Have a customer with a fairly large scale project that needs to get done, yesterday. Not sure how they thought they would be able to complete this internally, but they have basically a week or so to pull this off. Here is a list of requirements, if someone is interested in taking this on, preference would be for a single individual or firm to handle the job. Specifics are available, here is the overview. 1. We need an Engineer On-Call for a week for installation assistance (Asterisk Related) 2. Would like the Network Topology and Segmenting Double checked for accuracy and reliability. 3. Ensure proper connectivity with a Brook Trout Fax Server 4. Ensure interconnectivity with Modem Pool 5. Custom AGI script authoring MSSQL /mySQL to store call data 6. Custom script Training - What was done custom overview of why! 7. Setup of 3 T1 by working with CO 8. Custom Recording Application - files exported to NAS server w/naming convention to look-up calls i.e.800#,agentID,date,time 9. Someone available to work with us to ensure a smooth roll out. If anyone here feels they could handle this project, and can provide a few references please email me. Regards, -- Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC issues
I have been testing the ASTCC and have notice that when the caller hangs up the line while the balance is being played back the sub savedata() is not being called because the asterisk terminates the AGI and the rest of the script does not get executed thus never returning: AGI Script astcc.agi completed, returning 0 This leave the inuse set to 1 and the pin can not be used. I am using the lastest CVS HEAD asterisk-perl-0.08 Any comments -Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote:
This is not a siemens pbx problem you set the
pridialplan = to national and that adds a number to the outgoing call or
something just use
Pridialplan = local
prilocaldialplan = local
and it should work
no uuuaaa the same problem.. ring in the extension 100.
I tried to open the file kds again and now it showed me your configuration
:) don't know why it did not show me before
Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 17:31
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the
pbx. and all incomming calls go to 100. thats the problem i will try to
solve this.
It could potentially be both. I would look at your extensions.conf
first though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example
100
102 etc. but when i truy to go outside with the 9 before the call
rings in the first extensions (100). this is a asterisk problem? or
a pbx problem?
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--
.-
Pablo Allietti
LACNIC
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---end quoted text---
--
.-
Pablo Allietti
LACNIC
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[Asterisk-Users] Echo on SPA-3000 FXO
I've had an spa3k in service here at the house for a while now. After some initial wrangling, it's been working okay. I've had to reboot it a couple times and have noticed something rather annoying though. My setup is pretty simple and, dare I say, common. I have the SPA-3000 inline between my incoming POTS line and the internal house phone. It's setup to deliver all calls (from the outside or internal phones) to my asterisk (CVS HEAD) server. The FXO is my default outbound path and I have a VOIP provider as a secondary. After rebooting the SPA-3000, the internal users of calls routed through the FXO interface hear pretty bad echo. This persists for days, maybe more than a week. At some point, the echo goes away. I've noticed that, when the echo is gone, I hear a rapid series of light clicks on the line when placing a call; after dialing and before the remote end starts ringing. When the *is* echo, I'm not hearing the clicks. Seems to me these clicks are part of the echo training. How do I get this to occur immediately after the SPA-3000 restarts? Why do I have to wait so long to get them started? Paul -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA -- Onsite at GDOT W.Annex 404-463-2860 x199 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
