[asterisk-users] preventing voicemail pickup after SIP redirect ?
Hello, I'm using the classic [stdexten-macro] in extensions.conf whereby a call is picked up by voicemail after a certain ringing time. When programming a SIP phone to redirect calls (SIP 302 redirect) to another extension I'd like to avoid that voicemail pickup so that the call goes into the new destination's voicemail (if applicable). How can I detect that a call has been redirected and should no longer be intercepted by vm? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Faxing Support
Andrew Kohlsmith wrote: Undue? Digium requires disclaimers so they can dual-license it for ABE and other commercial vendors. You're purposely twisting and distorting the reality with these weasel words. I understand Digium strategy but I don't agree with it. I think it's wrong not to include code in Asterisk just because they won't be able to use it in ABE, so noncommercial version would be better. Asterisk isn't strong because of ABE and commercial installations, but because of big number of users and developers. Doing thing's like this Digium is pushing people away from Asterisk. If you don't like it, use something else. There's no need to take jabs at the company. You are not helping neither. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Digium cards on Vmware
Kevin P. Fleming wrote: The card manufacturer is irrelevant, as is the type of card. VMware does not currently provide any sort of PCI bus passthrough to virtual machines.. Hopefully this will change soon. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] web based sipphone
Hi dear is any web based sip-phone?opensource? best Mani Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA. http://answers.yahoo.com/dir/?link=listsid=396545367 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: FAX using T38
Steve Underwood wrote: I'll do it for 30% less than they quote. :-) I didn't see on their pages, what is their price? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Setting Sip Headers From Dial App?
SS == Stuart Sheldon [EMAIL PROTECTED] writes:
SS This might sound strange, but is there anyway for Asterisk to set
SS extra sip headers based on a sip phone returning a 302 in a
SS dialplan?
You can detect that a redirect has occurred by looking at ${RDNIS}.
You can't tell which SIP phone did the redirect though, which is
rather annoying -- you can't bill the redirected call to a specific
phone.
If it is an unattended transfer instead (I don't know if those use 302
as well), you can check ${BLINDTRANSFER}. That way you'll even know
which phone the transfer came from -- very useful for billing and
other purposes.
/Benny
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Micros-Fidelio - billing in hotel
There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something that is used world wide. Any informations are welcome. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server generated outbound conference calls?
I think you can do this with outlook. Use the Third Lane dialer product, set your extension to that of the conference, then initiate the calls. It will call the extension then the party and connect the two. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [asterisk_voip] asterisk and ogg files
Hello, Is it possible to use ogg stream with asterisk as moh? I have an icecast2 ogg streamer, but cannot use it with asterisk 1.4 The moh with files works icecast2 works but not icecast2+asterisk. I think I need something like (see below) in music on hold config file: mode=custom application=/usr/bin/some_player http://icecast_server:8000/mount Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
[EMAIL PROTECTED] wrote: I joined VirtualPhoneLine.Com service and am really enjoying the use of it. I am pretty certain this constitutes fraudulent and *misrepresentative* http://www.google.com/search?hl=ensa=Xoi=spellresnum=0ct=resultcd=1q=advertising+misrepresentativespell=1 advertising. You cannot represent yourself as a user and claim to have experiencing a product favourably if you own or are materially connected with that product. This action could attract the attention of the FTC and result in large fines for Mr. Rehan under the *Testimonials and Endorsements rules of internet advertising*. Connections between an endorser and the company that are unclear or unexpected to a customer also must be disclosed, whether they have to do with a financial arrangement for a favorable endorsement, a position with the company, or stock ownership. Expert endorsements must be based on appropriate tests or evaluations performed by people that have mastered the subject matter. /See/ FTC Guides Concerning Use of Endorsements and Testimonials in Advertising http://www.ftc.gov/bcp/guides/endorse.htm. http://www.ftc.gov/bcp/conline/pubs/buspubs/ruleroad.htm Of course, it would be highly unlikely anyone on the list would want to report Rehan...but in case anyone does: The FTC works for the consumer to prevent fraudulent, deceptive and unfair business practices in the marketplace and to provide information to help consumers spot, stop and avoid them. To file a complaint https://rn.ftc.gov/dod/wsolcq$.startup?Z_ORG_CODE=PU01 or to get free information on consumer issues http://www.ftc.gov/ftc/consumer.htm, visit _ www.ftc.gov http://www.ftc.gov_ or call toll-free, 1-877-FTC-HELP (1-877-382-4357); TTY: 1-866-653-4261. The FTC enters Internet, telemarketing, identity theft and other fraud-related complaints into Consumer Sentinel http://www.consumer.gov/sentinel, a secure, online database available to hundreds of civil and criminal law enforcement agencies in the U.S. and abroad. https://rn.ftc.gov/pls/dod/wsolcq$.startup?Z_ORG_CODE=PU01 -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium cards on Vmware
On 3/1/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Tomislav Parèina wrote: Is it possible to use Digium (or Sagnoma, or Beronet) cards with Asterisk on Vmware? The card manufacturer is irrelevant, as is the type of card. VMware does not currently provide any sort of PCI bus passthrough to virtual machines. Does anyone know if it will work with Xen? -- Morten Isaksen http://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium cards on Vmware
Morten Isaksen ha scritto: On 3/1/07, *Kevin P. Fleming* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Tomislav Parèina wrote: Is it possible to use Digium (or Sagnoma, or Beronet) cards with Asterisk on Vmware? The card manufacturer is irrelevant, as is the type of card. VMware does not currently provide any sort of PCI bus passthrough to virtual machines. Does anyone know if it will work with Xen? Tested and working on Xen, but only on the dom0 is usable. Even if it is possibile to use the pci bus from a DOMU. I tested a 4S0 board and in domu worked fine, but only with visdn, with misdn no way to reach somehing usable. The BIG problem is the unhappy visdn driver (i use a personal patched version working but incomplete and not stable so not to be placed in production). In the dom0 i found a near no-xen performance, no problems even with misdn so working great. I think the big problem is related to the poor realtime performance of XEN under load. :-) I think that the poor performance of Vmware is also a bottleneck so dont'think this can be a solution. Bye. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Using Asterisk as Voicemail Server on a dinosaurMeridian System
Use a Citel portico Telephone VoIP Adapter to interface the Meridian phones direct to the Asterisk server http://www.citel.com/. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J French Sent: 06 March 2007 00:04 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Using Asterisk as Voicemail Server on a dinosaurMeridian System We have a dinosaur Meridian system (~version 2) with 4 digital lines going to a Repartee Voicemail server. The Repartee got smoked by lightning two days ago and I'm itching to get Asterisk installed in its place. PRI is not an option since the system is so old that it doesn't even support PRI. I need to figure out how to connect the old Meridian to Asterisk otherwise. Any advice on getting Asterisk to work in its place is really appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] visdn, misdn and the hell
I am at the end of a long way... i try to work with a number of isdn boards (BRI not PRI) and i found only a lot of problems. First, the bristuff that is near working, but not so perfect ISDN designed interface. This is not bad but in a production environment this solution is not usable. Second visdn, i am cryng because i see a lot of people sayng yes is very attractive but the developer and who pay him is doing a bad bad bad bad work, they work on a closed source way, very bad. And last misnd, working now (thnks to Digium), i am happy, but not at 100%, a lot of problems are here, some kernel fault even, some problem in the diagnostic part. I think the ISDN part of asterisk is very important, in Italy there is a lot of equipments that are ISDN and not ANALOGIC or PRI, and with no ISDN stable support it is impossibile to port asterisk on the real world. Wath i see now is that a lot of integrators are doing this: using external box to avoid at 100% the isdn problem in asterisk. Very bad, we go to use proprietary designed hardware and software, external components, more complexity, more point of failure. This is, for me, the hell. I am the only one searching a solution to the ISDN problem? Why develop a lot separate ISDN systems and not to concentrate all efforts on one or two? Why there is NOT a native isdn support on asterisk? why i must add a separate visdn misdn bristuff or capi component? Ciao. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
The dialplan looks OK, depending of course on the numbers you trying to dial. If you want the phone to wait for a given timeout period after the digits are entered add a T immediately after the specific dialplan rule. (ie: xx[2-9]xxT). I'm assuming from your rules you need to dial a 9 first. Depending upon your numbering plan you could try ( 9x.T|0T|9011xxx.T ) -Steve »Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan from phone1.cfg: dialplan dialplan.1.impossibleMatchHandling=0 dialplan.1.removeEndOfDial=1 dialplan.2.impossibleMatchHandling=0 dialplan.2.removeEndOfDial=1 dialplan.3.impossibleMatchHandling=0 dialplan.3.removeEndOfDial=1 dialplan.4.impossibleMatchHandling=0 dialplan.4.removeEndOfDial=1 dialplan.5.impossibleMatchHandling=0 dialplan.5.removeEndOfDial=1 dialplan.6.impossibleMatchHandling=0 dialplan.6.removeEndOfDial=1 digitmap dialplan.1.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.1.digitmap.timeOut=3 dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan.3.digitmap.timeOut= dialplan.4.digitmap= dialplan.4.digitmap.timeOut= dialplan.5.digitmap= dialplan.5.digitmap.timeOut= dialplan.6.digitmap= dialplan.6.digitmap.timeOut=/ routing server dialplan.1.routing.server.1.address=10.0.17.8 dialplan.1.routing.server.1.port=5060 dialplan.2.routing.server.1.address= dialplan.2.routing.server.1.port= dialplan.3.routing.server.1.address= dialplan.3.routing.server.1.port= dialplan.4.routing.server.1.address= dialplan.4.routing.server.1.port= dialplan.5.routing.server.1.address= dialplan.5.routing.server.1.port= dialplan.6.routing.server.1.address= dialplan.6.routing.server.1.port=/ emergency dialplan.1.routing.emergency.1.value= dialplan.1.routing.emergency.1.server.1= dialplan.2.routing.emergency.1.value= dialplan.2.routing.emergency.1.server.1= dialplan.3.routing.emergency.1.value= dialplan.3.routing.emergency.1.server.1= dialplan.4.routing.emergency.1.value= dialplan.4.routing.emergency.1.server.1= dialplan.5.routing.emergency.1.value= dialplan.5.routing.emergency.1.server.1= dialplan.6.routing.emergency.1.value= dialplan.6.routing.emergency.1.server.1=/ /routing /dialplan Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 1.4 lost internet internal phones loose registration
Tomislav Parcina wrote: Thomas Kenyon wrote: Asterisk also seems to barf if it makes a registration/renewal request and it doesn't receive a reply in a timely fashion which will obviously happen if your internet connection disappears. (all versions I've used). That's why people should use dnsmasq. I used bind as the only nameserver in resolv.conf and it still happens. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
»Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ You're missing your pipes, also using a comma after a 9 will give a simulated second dial tone. digitmap=9[2-9]xx|[2-9]xx|9,1[2-9]xx|[2-9]xx The running count can be disabled by looking in the sip.cfg for: feature.8.enabled=1 Change it to a 0. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
Axel Thimm wrote: Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. They don't have 1.2.x version there? How fast do they make package since source version is out? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
Tomislav Parcina wrote: They don't have 1.2.x version there? Newer mind, I found it :) How fast do they make package since source version is out? This question still stands. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
DST rules can be found by searching the sip.cfgg file for SNTP. You will find a cluster of time parameters, including the month and day upon which to change DST. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Mar 5, 2007, at 9:20 PM, »Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9] xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan from phone1.cfg: dialplan dialplan.1.impossibleMatchHandling=0 dialplan. 1.removeEndOfDial=1 dialplan.2.impossibleMatchHandling=0 dialplan.2.removeEndOfDial=1 dialplan. 3.impossibleMatchHandling=0 dialplan.3.removeEndOfDial=1 dialplan.4.impossibleMatchHandling=0 dialplan. 4.removeEndOfDial=1 dialplan.5.impossibleMatchHandling=0 dialplan.5.removeEndOfDial=1 dialplan. 6.impossibleMatchHandling=0 dialplan.6.removeEndOfDial=1 digitmap dialplan.1.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx [2-9]xx dialplan.1.digitmap.timeOut=3 dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan. 3.digitmap.timeOut= dialplan.4.digitmap= dialplan. 4.digitmap.timeOut= dialplan.5.digitmap= dialplan. 5.digitmap.timeOut= dialplan.6.digitmap= dialplan. 6.digitmap.timeOut=/ routing server dialplan.1.routing.server.1.address=10.0.17.8 dialplan.1.routing.server.1.port=5060 dialplan.2.routing.server. 1.address= dialplan.2.routing.server.1.port= dialplan. 3.routing.server.1.address= dialplan.3.routing.server.1.port= dialplan.4.routing.server.1.address= dialplan.4.routing.server. 1.port= dialplan.5.routing.server.1.address= dialplan. 5.routing.server.1.port= dialplan.6.routing.server.1.address= dialplan.6.routing.server.1.port=/ emergency dialplan.1.routing.emergency.1.value= dialplan. 1.routing.emergency.1.server.1= dialplan.2.routing.emergency. 1.value= dialplan.2.routing.emergency.1.server.1= dialplan. 3.routing.emergency.1.value= dialplan.3.routing.emergency. 1.server.1= dialplan.4.routing.emergency.1.value= dialplan. 4.routing.emergency.1.server.1= dialplan.5.routing.emergency. 1.value= dialplan.5.routing.emergency.1.server.1= dialplan. 6.routing.emergency.1.value= dialplan.6.routing.emergency. 1.server.1=/ /routing /dialplan Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
Am Dienstag, den 06.03.2007, 05:18 -0400 schrieb Chris Mason (Lists): Of course, it would be highly unlikely anyone on the list would want to report Rehan...but in case anyone does: I have been told that unsolicited commercial e-mail (I do not imply that Rehan's post fulfills the criteria, judge yourself) may be * forwarded to [EMAIL PROTECTED] * for the people there to take further measures (whatever that means). I wonder that this fact is obviously widely unknown to both citizens and US-based spammers. I further wonder wether they rate the incoming complaints by number per incident, and perhaps prefer to prioritize the cases apparently more interesting to the wide public. I am not a US citizen, so FTC probably does not care for my 2c. Worth a try, anyway. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 501 - Auto answer on one line appearance
I am using SugarCRM together with the asterisk plugin, which allows me to click a number, SugarCRM calls my extension then places the call when I pickup. I would like to have that extension auto-answer. I set it up as line 3 on my phone so normal calls do not get auto-answered. However, I have not been able to get this to work. Has anyone implented this? This is what I put in the config file for the phone ringtype se.rt.enabled=1 se.rt.1.enabled=1 se.rt.1.ringer=9 se.rt.1.type=ring se.rt.2.enabled=1 se.rt.2.ringer=10 se.rt.2.type=ring se.rt.3.enabled=1 se.rt.3.ringer=11 se.rt.3.timeout=1000 se.rt.3.type=ring-answer se.rt.3.name=Ring Answer / -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent
Hey, Implementing Asterisk on local Lan spread over 2 campuses on two different cities is our graduation project. Having done all the research and reading stuff. I started with the practical work. Not getting a hand on the linux digium. I installed Red Hat linux 9.0. I was able to install zaptel1.4.0 and libpri1.4.0 successfully. When i attempted to install linux and put in following commands. # cd /usr/src/asterisk-version # make clean # make at the make command i got following error: The configure script must be make before running make make: *** [makeopts] Error 1 Please somebody could help me out. Regards, The Szabistians Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent
On Tue, 6 Mar 2007 12:22:24 + (GMT)
Asterisk Asterisk [EMAIL PROTECTED] wrote:
Hey,
Implementing Asterisk on local Lan spread over 2 campuses on two different
cities is our graduation project.
Having done all the research and reading stuff. I started with the
practical work.
Not getting a hand on the linux digium. I installed Red Hat linux 9.0. I
was able to install zaptel1.4.0 and
libpri1.4.0 successfully. When i attempted to install linux and put in
following commands.
# cd /usr/src/asterisk-version
# make clean
# make
at the make command i got following error:
The configure script must be make before running make
make: *** [makeopts] Error 1
Please somebody could help me out.
Regards,
The Szabistians
Send instant messages to your online friends http://uk.messenger.yahoo.com
./configure
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
// SIP:[EMAIL PROTECTED]
// Phone:+44 845 869 2749
--
This message has been scanned for viruses and dangerous content by MailScanner,
and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
On Tue, Mar 06, 2007 at 11:59:10AM +0100, Tomislav Parcina wrote: Tomislav Parcina wrote: They don't have 1.2.x version there? Newer mind, I found it :) How fast do they make package since source version is out? This question still stands. As fast as they read asterisk-announce ;) -- Axel.Thimm at ATrpms.net pgpkcoggk14kl.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing does not terminate on mISDN after pickup
Hello, I am having something of an odd problem: about every 100 calls or so, when a call comes in via an external mISDN interface and I route it to an internal mISDN interface by dialing an internal msn that is programmed for multiple phones on the internal bus, somtimes the other phones continue ringing for several minutes after the call has already been picked up by one (or even eventually hungup already...). As you imagine this is really annoying... I don't seem to be able to narrow it down in any way: show channels is empty (provided, call was answered an hungup, other phones continue ringing) The only way to terminate the ringing is: wait for several minutes... :-) or: restart asterisk. Can anybody point me in the right direction with this issue? Every attempt to get rid of the problem has failed. I have no clue what else I could try... - Does anybody else experience this as well? Cheers, Arik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] preventing voicemail pickup after SIP redirect ?
Louis-David Mitterrand wrote: Hello, I'm using the classic [stdexten-macro] in extensions.conf whereby a call is picked up by voicemail after a certain ringing time. When programming a SIP phone to redirect calls (SIP 302 redirect) to another extension I'd like to avoid that voicemail pickup so that the call goes into the new destination's voicemail (if applicable). How can I detect that a call has been redirected and should no longer be intercepted by vm? That should happen by default. The call should get sent to the new place and it should act like the call was directly dialed to that extension. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
Hi,guys. I try to install zaptel 1.4 on dell 2850, my OS is fc6, I can compile and install it with no error. But when I modprobe wd4xxp module, the OS hang. pI must push the reset button on 2850's panel to reboot the OS. but when OS starting udev, it can't continue, that is, I can't enter the OS any more.ppAny one can help me? thank you.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
Here is some of my actual Polycom config files. The only thing that has been changed is the hostname of the PBX. We assign the SIP user ID as the MAC of the phone with a -a -b -c, etc appended to it for each of the line appearances. These config files do not have the DST stuff added yet. http://www.fnords.org/~eric/polycom-config-examples/ Steve Blair wrote: The dialplan looks OK, depending of course on the numbers you trying to dial. If you want the phone to wait for a given timeout period after the digits are entered add a T immediately after the specific dialplan rule. (ie: xx[2-9]xxT). I'm assuming from your rules you need to dial a 9 first. Depending upon your numbering plan you could try ( 9x.T|0T|9011xxx.T ) -Steve »Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan from phone1.cfg: dialplan dialplan.1.impossibleMatchHandling=0 dialplan.1.removeEndOfDial=1 dialplan.2.impossibleMatchHandling=0 dialplan.2.removeEndOfDial=1 dialplan.3.impossibleMatchHandling=0 dialplan.3.removeEndOfDial=1 dialplan.4.impossibleMatchHandling=0 dialplan.4.removeEndOfDial=1 dialplan.5.impossibleMatchHandling=0 dialplan.5.removeEndOfDial=1 dialplan.6.impossibleMatchHandling=0 dialplan.6.removeEndOfDial=1 digitmap dialplan.1.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.1.digitmap.timeOut=3 dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan.3.digitmap.timeOut= dialplan.4.digitmap= dialplan.4.digitmap.timeOut= dialplan.5.digitmap= dialplan.5.digitmap.timeOut= dialplan.6.digitmap= dialplan.6.digitmap.timeOut=/ routing server dialplan.1.routing.server.1.address=10.0.17.8 dialplan.1.routing.server.1.port=5060 dialplan.2.routing.server.1.address= dialplan.2.routing.server.1.port= dialplan.3.routing.server.1.address= dialplan.3.routing.server.1.port= dialplan.4.routing.server.1.address= dialplan.4.routing.server.1.port= dialplan.5.routing.server.1.address= dialplan.5.routing.server.1.port= dialplan.6.routing.server.1.address= dialplan.6.routing.server.1.port=/ emergency dialplan.1.routing.emergency.1.value= dialplan.1.routing.emergency.1.server.1= dialplan.2.routing.emergency.1.value= dialplan.2.routing.emergency.1.server.1= dialplan.3.routing.emergency.1.value= dialplan.3.routing.emergency.1.server.1= dialplan.4.routing.emergency.1.value= dialplan.4.routing.emergency.1.server.1= dialplan.5.routing.emergency.1.value= dialplan.5.routing.emergency.1.server.1= dialplan.6.routing.emergency.1.value= dialplan.6.routing.emergency.1.server.1=/ /routing /dialplan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup failover
hello,
i have configured internal pickup my problem is when an external call is
coming i cannot pickup coz the extension is an external number.
isit possible to pickup the external via n+101 prio or is ther any other
solution?
my config:
exten = _*8.,1,GoToIf($[${CDR(userfield)} = EXTERN_INCOMING]?10)
exten = _*8.,2,Pickup(${EXTEN:[EMAIL PROTECTED])
regards rene
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: TDM400P/FXS in a HP DL380 G5
James FitzGibbon wrote: They say it's custom made for them, and I certainly can't find anything else like it after several hours of searching, but it seems to be what's required. I'll have to rig up a backplate with a cutout to get the 12V connector into the case, but other than that I'm hoping it will do the trick. Contact the Digium Sales department... very shortly we will have a product called the PWR2400B which combines a backplane bracket, one (or two) 12V power supplies and the appropriate internal wiring to power any cards that need a 12V power source. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g.729 on solaris10/x86
Juraj Bednar wrote: I'm looking for a way to have G.729 codec working on Solaris/x86. Binary codec from Digium is not compiled for Solaris/x86 (only sparc). Are there any alternative (free or commercial) G.729 implementations, which would work? We will have Solaris 10 x86-32 and x86-64 binaries of the G.729 codec and registration tool in our 'unsupported' directory later this week. There will be an announcement on the lists when this happens, so stay tuned. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???
Mr. James W. Laferriere wrote: Hello All , I'd usually just take the latest timestamped tarballs use them , But this has gotten me a tad setback . I want to build astersik-1.4.1 I am not sure which of these is going to work correctly . Anyone else have a better idea than me ? You are comparing extra-sounds and core-sounds, which are different packages with separate version number sequences. They are completely unrelated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: preventing voicemail pickup after SIP redirect ?
On Tue, Mar 06, 2007 at 07:18:08AM -0600, Eric ManxPower Wieling wrote: How can I detect that a call has been redirected and should no longer be intercepted by vm? That should happen by default. The call should get sent to the new place and it should act like the call was directly dialed to that extension. Actually no. When a call coming in through Zap, Capi or mISDN is redirected by a SIP phone with a 302, then asterisk creates a Local/xx channel to the new destination, while the original channel is still open. So after $RINGTIME is reached, [stdexten-macro] answers the original call and sends it to the original extension's vm. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED
On Mar 6, 2007, at 1:55 AM, Tomislav Parcina wrote: Kristian Kielhofner wrote: Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Hi Kristian! Thank you for your work. I'm not able to test this right now, but I'll sourly need this sometimes. Hmmm - might be me but I am unable to find the beginning of this thread. It does sound interesting. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_cellphone won't pair with phone
I'm running chan_cellphone version 13 on the latest svn trunk (as
root). I believe I have chan_cellphone set up correctly (bt addr and
port retrieved from the cell search CLI command). When I load the
chan_cellphone module, my Motorola V3m asks if I want to allow Asterisk
PBX, I say yes and enter the for the pin, then my phone tells me
the pin is invalid. Here is the log entry generated by chan_cellphone:
*CLI [Mar 3 22:53:12] DEBUG[16888]: chan_cellphone.c:682
rfcomm_connect: connect() failed (111).
*CLI cell show devices
ID Address Connected State
razrxx:xx:xx:xx:xx:xx NoInit
I'm running:
- Fedora Core 6 - kernel 2.6.19-1.2911.6.4.fc6
- bluez-libs (libbluetooth) 3.7-1
- bluez-utils 3.7-2
I'm using the hcid.conf and pinhelper script from contrib/bluetooth
(which I moved to /etc/bluetooth).
I'm not sure how to debug this further. Any ideas?
Earle
---
Here's the directory listing and the contents of the files (comments
removed for brevity):
[EMAIL PROTECTED] ~]# ls -l /etc/bluetooth/
total 12
-rw-r--r-- 1 root root 1428 Mar 3 18:36 hcid.conf
-rwxr-xr-x 1 root root 27 Mar 3 18:36 pinhelper
-rw-r--r-- 1 root root 297 Oct 2 18:40 rfcomm.conf
--- pinhelper ---
#!/bin/sh
echo PIN:
--- hcid.conf ---
options {
autoinit yes;
security auto;
pairing multi;
pin_helper /etc/bluetooth/pinhelper;
}
device {
name Asterisk PBX;
class 0x3e0100;
iscan enable; pscan enable;
lm accept;
lp rswitch,hold,sniff,park;
auth enable;
encrypt enable;
}
--- cellphone.conf ---
[general]
interval=60; Number of seconds between trying to connect to
devices.
[razr]
address=xx:xx:xx:xx:xx:xx ; retrieved from cell search CLI command
port=4
context=incoming-mobile
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
On Tue, Mar 06, 2007 at 09:22:44PM +0800, [EMAIL PROTECTED] wrote: Hi,guys. I try to install zaptel 1.4 on dell 2850, my OS is fc6, I can compile and install it with no error. But when I modprobe wd4xxp wc4xxp ? module, the OS hang. I must push the reset button on 2850's panel to reboot the OS. but when OS starting udev, it can't continue, that is, I can't enter the OS any more. What kernel version exactly? uname -a What version of zaptel do you use? modinfo zaptel | grep ^version: -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys PAP2 and Caller ID
Hi! Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to show the Caller number on the phone. There's a Caller ID Method: option on Regional settings, but I tested all options, and my CLIP phone never shows the Caller number... :( Any idea? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
»Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan snip Polycom's 2.1.0 firmware has the new DST settings as the default. This is what they use for the SNTP element: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ The one thing I'm not sure about is the tcpIpApp.sntp.daylightSavings.start.date=8 line. According to the 2.1.0 admin guide that means the second week of the month but none of the guides before that mention this as a valid option. The missed calls option can be enabled/disabled by changing the feature element. Specifically feature.8.enabled=0. This will disable both the message and the ability to see missed calls. Your dialplan looks syntactically correct but only allows for 10 digit phone numbers. Perhaps you want something more like this? dialplan.digitmap=9,[2-9]xx|9,1[2-9]x Hope that helps -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits sent on IAX native bridge
Any ideas as to how I can fix this issue? Thanks Remi Remi Quezada wrote: Ok that makes sense, but I'm still getting double digits. It seems to me that the DTMF digit is getting detected too late. When the digit is pressed it seems like asterisk is passing the DTMF digit for a fraction of a second through the audio path and then sends the digit for however long your toneduration is set for. I can hear this happening when I dial the digits myself, I hear some kind sound being cut off for a fraction of a second and then hear the DTMF tone pass. So I guess this is why sometimes some answer machines are detecting double digits. Russell Bryant wrote: Remi Quezada wrote: I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable debugging for iax and I do see it sending the DTMF digits two times. Here is what I see: The IAX debug that you show below only shows one of each digit. For each one, it shows Receiving the digit from one leg of the call, and then transmitting it out the other. I have spaced out your debug to separate each digit. Each one shows ... - digit - ACK - - digit --- ACK -- which is exactly what is supposed to happen. Rx-Frame Retry[ No] -- OSeqno: 018 ISeqno: 021 Type: DTMFSubclass: 1 Timestamp: 51523ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 019 Type: IAX Subclass: ACK Timestamp: 51523ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 019 ISeqno: 022 Type: DTMFSubclass: 1 Timestamp: 51543ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 020 Type: IAX Subclass: ACK Timestamp: 51543ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 019 ISeqno: 021 Type: DTMFSubclass: 2 Timestamp: 52083ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 020 Type: IAX Subclass: ACK Timestamp: 52083ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 020 ISeqno: 022 Type: DTMFSubclass: 2 Timestamp: 52103ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: IAX Subclass: ACK Timestamp: 52103ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 020 ISeqno: 021 Type: DTMFSubclass: 3 Timestamp: 52663ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 021 Type: IAX Subclass: ACK Timestamp: 52663ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 021 ISeqno: 022 Type: DTMFSubclass: 3 Timestamp: 52683ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 022 Type: IAX Subclass: ACK Timestamp: 52683ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 021 ISeqno: 021 Type: DTMFSubclass: 4 Timestamp: 53223ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 022 Type: IAX Subclass: ACK Timestamp: 53223ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 022 ISeqno: 022 Type: DTMFSubclass: 4 Timestamp: 53243ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 023 Type: IAX Subclass: ACK Timestamp: 53243ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: DTMFSubclass: 5 Timestamp: 53703ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 023 Type: IAX Subclass: ACK Timestamp: 53703ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 023 ISeqno: 022 Type: DTMFSubclass: 5 Timestamp: 53723ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 024 Type: IAX Subclass: ACK Timestamp: 53723ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 023 ISeqno: 021 Type: DTMFSubclass: 6 Timestamp: 54163ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 024 Type: IAX Subclass: ACK Timestamp: 54163ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 024 ISeqno: 022
[asterisk-users] Asterisk 1.2.15 chan_vpb with vpb-driver 4.0
Hi, all, I am using Asterisk 1.2.15 with an OpenLine4 card (vpb-driver 4.0). And Asterisk segfaults. Here is the output of loading chan_vpb. Very detailed because I turned on vpb verbose. any lead to solution will be appreciated. Thanks output from Asterisk: [chan_vpb.so] = (VoiceTronix V6PCI/V12PCI/V4PCI API Support) == Parsing '/etc/asterisk/vpb.conf': Found Mar 6 15:22:00 NOTICE[12913]: chan_vpb.c:2809 int load_module(): VPB Driver configured to use [1] cards We have a valid config file combination... Number of OpenPri cards: 0 Number of OpenSwitch cards: 0 Number of OpenLine cards: 1 Number of vtcore cards: 0 Sum Total number of Cards: 1 number of cards [1] Card Number = 0 Model = 2 (from config file) defRecordGain=6.00 defPlayGain=7.00 Initialies VPB Registry, located [1] boards Initializing board[0][0] model[2] About to load firmware... DSP [00] Memory test passed OK... [0]: .dspint flags: 0x0040 addr: 0x length: 0x00a8 . [1]:.text flags: 0x0020 addr: 0x0200 length: 0x61ec . [2]: .cinit flags: 0x0050 addr: 0x0200 length: 0x0320 ... [3]: .switch flags: 0x0040 addr: 0x63ec length: 0x00ac . [4]:.ebss flags: 0x0080 addr: 0x6600 length: 0x007f [5]:.data flags: 0x0040 addr: 0x667f length: 0x02ef . [6]: .bss flags: 0x1080 addr: 0x6980 length: 0x0426 [7]: .const flags: 0x0040 addr: 0x6da6 length: 0x0bc1 . [8]: .stack flags: 0x0080 addr: 0x7b00 length: 0x0c00 [9]: .sysmem flags: 0x0080 addr: 0x8700 length: 0x7800 [10]: .b0data flags: 0x0080 addr: 0x0100 length: 0x0090 [11]: .b0prog flags: 0x0080 addr: 0xfe00 length: 0x0090 [12]: .b1data flags: 0x0080 addr: 0x0300 length: 0x0090 [13]: .B1 flags: 0x0080 addr: 0x0300 length: 0x0174 [14]: .b2data flags: 0x addr: 0x0060 length: 0x [15]: .B2 flags: 0x0080 addr: 0x0060 length: 0x0017 [16]:.init flags: 0x0040 addr: 0x length: 0x Entry point: 0x0200 DSP [00] code downloaded OK About to run... DSP running... model = 0 vr[i].model = 2 DSP [00] Message FIFOs booted OK Generic_pci_block_eeread: data [0xbffc9fb8] Configuring [1] VPBs... Opening Tone detector for [4] channels... Configuring each channel device... dev: 0 :Down channel objects, obj:11 dev: 0 :call Progress state machines dev: 1 :Down channel objects, obj:31 dev: 1 :call Progress state machines dev: 2 :Down channel objects, obj:51 dev: 2 :call Progress state machines dev: 3 :Down channel objects, obj:71 dev: 3 :call Progress state machines Starting config manager for board [0].. Taking a nap to let the board start up override default prog tones from reg VPBs configured OK! Opening PlayRec module... playrec.cpp: setting V20.03 hyb bals on OL playrec.cpp: setting V20.03 hyb bals on OL playrec.cpp: setting V20.03 hyb bals on OL playrec.cpp: setting V20.03 hyb bals on OL Starting loop sense thread monitoring on [4] channels Loop Sense Thread started OK! timer callback started OK! Driver initialised OK! Segmentation fault , -- /* * Yifan Zhang * * Softsound * +44 (0)1223 448 021 */ Vim is the best editor in the world! - C Programmer With Cream, it is even better! - C Programmer programming in Java The information contained in this message is for the intended addressee only and may contain confidential and/or privileged information. If you are not the intended addressee, please delete this message and notify the sender, and do not copy or distribute this message or disclose its contents to anyone. Any views or opinions expressed in this message are those of the author and do not necessarily represent those of Autonomy Systems Limited or of any of its associated companies. No reliance may be placed on this message without written confirmation from an authorised representative of the company. Autonomy Systems Limited, Registered Office: Cambridge Business Park, Cowley Road, Cambridge CB3 0WZ, Registered Number 03063054. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:Problem with TE212P
Hello This is another method, if you don want to change zaptel.conf (from Chris Hozian of Digium) I would like to clarify that there is another method which may be used that does not require you to load the kernel modules in a different order or to modify your zaptel.conf and zapata.conf file. This can be done by specifying trunkgroup and spanmaps in your zapata.conf file. [trunkgroups] trunkgroup = 1,40 trunkgroup = 2,71 spanmap = 1,1 spanmap = 2,2 Best Reggards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
In 1.2, try adding noload = res_musiconhold.so to your modules.conf. In 1.4 though, it would be worth a try, but I don't know for sure if that's how it's done. Moj David Thomas wrote: On 3/5/07, C F [EMAIL PROTECTED] wrote: Could be its trying but does it actualy play the music? It's not actually playing anything. I guess it just seems odd that Asterisk re-invites the media back to itself when a call is put on hold (when MOH is disabled), instead of simply disconnecting the media until the call is retrieved. I guess I was hoping for a config option that would simply turn MOH off to achieve this behavior. Does such a config option exist in 1.4? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: queue information into db
i've submittet the project to SF. I have to wait 2 business days for their validation. The project is in a beta release and will be released on GPL. Bye On 3/2/07, Tomislav Parcina [EMAIL PROTECTED] wrote: nik600 wrote: i'm sorry but due to some problem the software will be released not first than Wednesday 7/02/2007. i'll post a message . This should be Wednesday 7/3/2007. right? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 - SLA
Bill Gibbs wrote: Sorry to reply to myself, once again onn the list, but since SLA is new I figured I should answer my own question before anyone else gets confused...I completely forgot about my -directory.xml defaults...so that's where all these bogus SUBSCRIBE requests were coming from. It's fine, please respond with your progress. I am eager to hear how it goes. In fact, I would like to expand the documentation to describe exactly how to set up specific phones to work properly in this setup. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
Doug Lytle wrote: »Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ You're missing your pipes, also using a comma after a 9 will give a simulated second dial tone. digitmap=9[2-9]xx|[2-9]xx|9,1[2-9]xx|[2-9]xx Actually, the pipes in the example above are correct. I want to be able to dial: 9nxxnxx and 91nxxnxx The running count can be disabled by looking in the sip.cfg for: feature.8.enabled=1 Change it to a 0. Thanks. This worked like a charm! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 - SLA
Bill Gibbs wrote: I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. If you see no subscriptions, then the phones will not dispaly the state of the line at all. In regards to still allowing you to dial when all lines are busy, do you have your phones set up to automatically dial when you go off-hook? In this SLA setup, you should not allow any dialing on the phone before a call is made. If the phone is taken off hook without pressing a specific line button, the phone should immediately dial the station1 (or whatever the station is named) extension. This will connect the station to the first available trunk if there is one, and then provide dialtone for making a call. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 - SLA
On 3/6/07, Russell Bryant [EMAIL PROTECTED] wrote: This will connect the station to the first available trunk if there is one, and then provide dialtone for making a call. That's what I was concerned about. Whether it connects to the first available, or the first one. In other words, if line 1 is in use, does it connect to line 1 or line 2? I haven't got a chance to get back on this yet. Hopefully I will be able to this afternoon and tomorrow. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Building a new voicemail system... Testers needed!
Friends in the Asterisk community, One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I've constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I've been kind of avoiding during my trainings is voicemail. And there's where I've spent a lot of time recently... Life is strange. Instead of fixing the current voicemail, I decided to restart. Break up large apps into small building blocks, allowing Asterisk admins to use the rich dialplan script language or AEL to build a voicemail solution that fits the organization. I've named this minivoicemail, which for each addition becomes more of a bad choice of name for this project. Flexivoicemail could be better... :-) I've removed functionality like ODBC and IMAP support, something that can be reapplied later. I've also not replaced the hooks into other channels for voicemail notification, but that can be done too. I haven't started replacing voicemailmain(), since I've focused on the need of larger systems where one only supports e-mail notifications of voicemail with audio attached. What I currently have is: Applications - MinivmGreet Play voicemail greetings (busy/unavailable/temporary) - MinivmRecord Record voicemail message - MinivmNotify Notify account owner of message (email, pager) - MinivmDelete Delete message Functions - MINIVMACCOUNT() - Get properties of voicemail account CLI commands - minivm show settings - minivm reload - minivm show stats - minivm list accounts - minivm list templates New features: - I've added support for e-mail and pager templates in various languages. - All apps are usable without setting up a voicemail account for a user. Just run the app with an e-mail address as an argument. The branch is based on Asterisk 1.2 and can easily be downloaded from http://svn.digium.com/svn/asterisk/team/oej/minivoicemail I need testers, ideas for new applications and possibly coders that can help to complete this. To start - Checkout this branch, compile and install - Check the minivm.conf.sample for instructions - Read the top of the source code file for ideas, todo's and changes http://svn.digium.com/view/asterisk/team/oej/minivoicemail/apps/ app_minivm.c?view=markup (And if you want to encourage me further, paypal to [EMAIL PROTECTED], thanks) Thanks for your help building a more flexible voicemail system for Asterisk! Send bug reports, comments and ideas directly to me and I'll try to summarize. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
I think it has something to do with hints...I can't seem to subscribe to anything now with 1.4 vs 1.2, even with a normal non SLA setup. My phone/config that works with 1.2, so I know hints work with the phone and firmware and with NAT at least on 1.2. I did a fresh 1.4 install (and I did a make samples so I had something to work off of) sip show subscriptions shows 0 active show hints: [EMAIL PROTECTED] : SIP/2404366402State:Idle Watchers 0 If I run the default demo app, show hints still shows Idle. My Buddies key in the Polycom, which is watching the proper sip hint (works in 1.2) shows the extension to be Offline. Sip.conf [general] allowsubscribe=yes subscribecontext=default notifyringing=yes notifyhold=yes limitonpeers=yes (I tried with and without the above values commented out, as well as specifically in my device peer definition) [2404366402] type=friend secret=blahededah nat=yes host=dynamic canreinvite=no context=default qualify=yes extensions.conf [default] exten = 2404366402,hint,SIP/2404366402 ...etc... My mac-directory.xml ..snip... item lnmyself/ln fn/fn ct2404366402/ct sd/sd rt/rt dc/ ad0/ad ar0/ar bw1/bw bb0/bb /item ...snip... I also tried in the ct[EMAIL PROTECTED]/ct Let's pretend 1.1.1.1 is my firewall that the Polycom is behind 2.2.2.2 is my 1.4.1 test Asterisk server --- SIP read from 1.1.1.1:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK256e271aEC1CEA7B From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - --- (13 headers 0 lines) --- Creating new subscription Sending to 192.168.1.116 : 5060 (no NAT) Found peer '2404366402' --- Transmitting (NAT) to 1.1.1.1:60671 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK256e271aEC1CEA7B;received=1.1.1.1 From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED];tag=as3123a96d Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7a544b2b Content-Length: 0 --- SIP read from 1.1.1.1:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK3f8d777548EC8ED2 From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED] CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Authorization: Digest username=2404366402, realm=asterisk, nonce=7a544b2b, uri=sip:[EMAIL PROTECTED]:5060, response=404b224f5abbdc3793d4df45ee2ffa59, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - --- (14 headers 0 lines) --- Creating new subscription Sending to 1.1.1.1 : 60671 (NAT) Found peer '2404366402' Looking for 2404366402 in default (domain 2.2.2.2) --- Transmitting (NAT) to 1.1.1.1:60671 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK3f8d777548EC8ED2;received=1.1.1.1 From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED];tag=as3123a96d Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Tuesday, March 06, 2007 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Bill Gibbs wrote: I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. If you see no subscriptions, then the phones will not dispaly the state of the line at all. In regards to still allowing you to dial when all lines are busy, do you have your phones set up to automatically dial when you go off-hook? In this SLA setup, you should not allow any
Re: [asterisk-users] running asterisk through cellphone
I've used the Dock 'n Talk, and I can say that it worked as well for us as it claimed to be able to. Only need an analog Zaptel card of some sort. I know there are a few other brands available, as well as some GSM Bridges available that you insert the SIM card directly into, bypassing the cellphone. Moj Gordon Henderson wrote: On Tue, 27 Feb 2007, Michael Kamleitner wrote: hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via data-cable (or bluetooth?) and use this as the single line to call-in. searching the asterisk-forums I found mentions of chan_cellphone, which is probably a patch for exactly this kind of usage, right? I'ld be thankful if you could just point me to the right direction (I'm quite new to asterisk). thx in advance! If I understand you, you want to call the mobile phone, and have asterisk deal with the audio? the only think I know of is Dock'n'Talk http://www.phonelabs.com/prd05.asp but that has an analogue output, so you'd need analogue into asterisk, and if you have analogue in, then you might as well use a landline if you can... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
I assume my SUBSCRIBE issue for hints has something to do with this bug http://bugs.digium.com/view.php?id=9168 Bill snipped previous emails for readability ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many gsm channels
Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked manager.conf (only lines are [general] enabled=yes). I've cleaned out /var/lib/asterisk. My full log shows the following every attempt: [Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Challenge' [Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Login' [Mar 6 13:32:39] VERBOSE[28578] logger.c: == Parsing '/etc/asterisk/manager.conf': [Mar 6 13:32:29] VERBOSE[28567] logger.c: Found [Mar 6 13:32:40] VERBOSE[28578] logger.c: == Connect attempt from '127.0.0.1' unable to authenticate I've updated from 1.2.13 to 1.4.1 and done everything I could to remove Trixbox from the picture. I thought for sure it was a module, but moving them all out of the picture didn't alleviate the problem. It seems as long as manager.conf exists I'm getting these messages. I've got 3 boxes setup with mostly identical setups (extensions.conf is different) and only one box is getting this message. From what I can tell from google searches it appears astbill and/or trixbox are likely to blame but I'm running out of places to look for these culprits. Any suggestinos would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
Hi Ken, Trixbox comes with the Flash Operator Panel. The FOP server is likely setup with incorrect authentication parameters, and hence is failing authentication everytime it attempts to use the Asterisk Manager API to update it's tracking of what's going on in your system. Check your op_server.cfg file (/var/www/html/panel/, I think). Look for the manager_user and manager_secret parameters, and make sure they match an entry in /etc/asterisk/manager.conf. Alex On 3/6/07, Ken Williams [EMAIL PROTECTED] wrote: Every few seconds I get the following message: * == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate * I'm trying to track down where it's coming from. I've used TCPDUMP NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked manager.conf (only lines are [general] enabled=yes). I've cleaned out /var/lib/asterisk. My full log shows the following every attempt: *[Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Challenge' [Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Login' [Mar 6 13:32:39] VERBOSE[28578] logger.c: == Parsing '/etc/asterisk/manager.conf': [Mar 6 13:32:29] VERBOSE[28567] logger.c: Found [Mar 6 13:32:40] VERBOSE[28578] logger.c: == Connect attempt from ' 127.0.0.1' unable to authenticate * I've updated from 1.2.13 to 1.4.1 and done everything I could to remove Trixbox from the picture. I thought for sure it was a module, but moving them all out of the picture didn't alleviate the problem. It seems as long as manager.conf exists I'm getting these messages. I've got 3 boxes setup with mostly identical setups (extensions.conf is different) and only one box is getting this message. From what I can tell from google searches it appears astbill and/or trixbox are likely to blame but I'm running out of places to look for these culprits. Any suggestinos would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How many gsm channels
We installed a quad xeon 3ghz which transcoded ~100 active channels (as a gateway). Take a look at the codec demands (in asterisk show codecs I believe) and scale from there. This box was 60% loaded - which is all we're comfortable with before latency goes too high. Michelle Dupuis Technical Support Specialist Generation Software - Linux and Asterisk solutions and support. Visit us at www.generationd.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Tuesday, March 06, 2007 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How many gsm channels Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
That was indeed the problem. I thought I had eliminated any httpd by disabling the service, but the problem was trixbox was still trying to load on startup via /usr/sbin/amportal. Once I removed that from startup, problem resolved. I did go a step further and wipe out the panel in httpd as well. Thanks for the pointer. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar Sent: Tuesday, March 06, 2007 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate' Hi Ken, Trixbox comes with the Flash Operator Panel. The FOP server is likely setup with incorrect authentication parameters, and hence is failing authentication everytime it attempts to use the Asterisk Manager API to update it's tracking of what's going on in your system. Check your op_server.cfg file (/var/www/html/panel/, I think). Look for the manager_user and manager_secret parameters, and make sure they match an entry in /etc/asterisk/manager.conf. Alex On 3/6/07, Ken Williams [EMAIL PROTECTED] wrote: Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked manager.conf (only lines are [general] enabled=yes). I've cleaned out /var/lib/asterisk. My full log shows the following every attempt: [Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Challenge' [Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Login' [Mar 6 13:32:39] VERBOSE[28578] logger.c: == Parsing '/etc/asterisk/manager.conf': [Mar 6 13:32:29] VERBOSE[28567] logger.c: Found [Mar 6 13:32:40] VERBOSE[28578] logger.c: == Connect attempt from '127.0.0.1' unable to authenticate I've updated from 1.2.13 to 1.4.1 and done everything I could to remove Trixbox from the picture. I thought for sure it was a module, but moving them all out of the picture didn't alleviate the problem. It seems as long as manager.conf exists I'm getting these messages. I've got 3 boxes setup with mostly identical setups (extensions.conf is different) and only one box is getting this message. From what I can tell from google searches it appears astbill and/or trixbox are likely to blame but I'm running out of places to look for these culprits. Any suggestinos would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
On Tue, Mar 06, 2007 at 02:26:21PM -0700, Ken Williams wrote: That was indeed the problem. I thought I had eliminated any httpd by disabling the service, but the problem was trixbox was still trying to load on startup via /usr/sbin/amportal. Once I removed that from startup, problem resolved. The authentication attempt came from the op_server.pl process, not from the remote client. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling smsq in 1.2
How to compile smsq in 1.2? It is compile in 1.4 by default. It is included in 1.2.13, but not compiled. Any rule or method to make it? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add current caller to junk-callers-database
Hello, I'm wondering how one might set up a feature to add (in real-time) the current CallerID information to a junk-callers database. After answering a call from an outside line and determining that the call was from a telemarketer or the like, the user could dial an easy specific code (like ** or 77), which would cause the call to be transferred to a specific extension within the Asterisk dialplan, where the CallerID info would be added to the database, a recorded message played to the caller, and then the call terminated. When that CallerID phoned again, the call would be diverted to voicemail or whatever automatically (instead of ringing). Another specific extension could be used to add/delete entries from the database if necessary. Any suggestion on how to enable a feature code to do the initial transfer? Thanks, Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cancelling a digit in IVR
Hello, I'm wondering how to make it possible for the user to cancel the last entered digit, if he made a mistake. For example, a user calls and starts entering 1...2...4, then he should be able to press, lets say *, to cancel 4 and enter i.e. 3. Thanks Jake -- --- Domeny w ULTRA NISKICH cenach: --- .pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl http://www.domeny.alpha.pl -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cancelling a digit in IVR
Why not just push * to start again. The customer most of the time wont know that they made a mistake until you are reading the digits back to them anyway. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kuba Sent: Tuesday, 6 March 2007 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cancelling a digit in IVR Hello, I'm wondering how to make it possible for the user to cancel the last entered digit, if he made a mistake. For example, a user calls and starts entering 1...2...4, then he should be able to press, lets say *, to cancel 4 and enter i.e. 3. Thanks Jake -- --- Domeny w ULTRA NISKICH cenach: --- .pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl http://www.domeny.alpha.pl -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: build rpm fails
Will try that again -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Tuesday, March 06, 2007 11:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: build rpm fails Axel Thimm wrote: Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. They don't have 1.2.x version there? How fast do they make package since source version is out? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Invoice Harry.MDI Description: image/vnd.ms-modi Invoice Antivirus.MDI Description: image/vnd.ms-modi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 - SLA
Lacy Moore - Aspendora wrote: That's what I was concerned about. Whether it connects to the first available, or the first one. In other words, if line 1 is in use, does it connect to line 1 or line 2? If you take a phone off hook without pressing a line button, and the phone is properly configured to initiate a call to the station1 extension, or equivalent, then it will choose the first available line. It will choose them in the order that they are specified in sla.conf. Of course, if you press a specific line button, it will connect to that line, regardless of whether it is in use or not. This behavior can be controlled by setting barge=no in sla.conf. I haven't got a chance to get back on this yet. Hopefully I will be able to this afternoon and tomorrow. It's all good. :) -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Questions
Dave Fullerton wrote on 3/6/07 9:33 AM: Polycom's 2.1.0 firmware has the new DST settings as the default. This is what they use for the SNTP element: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ The one thing I'm not sure about is the tcpIpApp.sntp.daylightSavings.start.date=8 line. According to the 2.1.0 admin guide that means the second week of the month but none of the guides before that mention this as a valid option. Thanks! One question I have... with this applied (and even with the original config I had before changing it to this), the start.dayOfWeek setting shows up as Monday on the web interface on the phone. Is the web interface goofed up, or should that be Sunday? -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
uname -a Linux xxx 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:37:32 EDT 2006 i686 i686 i386 GNU/Linux modinfo zaptel | grep ^version: version:1.4.0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
I can run zaptel 1.4 normally in other machine on the same OS, only can't run it on 2850. It hangs the OS.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Polycom reject button
exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten = 111,4,Goto(111-${DIALSTATUS},1)
exten = 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten = 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
This is awesome, I had actually wondered about doing this same thing
and never thought to ask about it. Thanks for sharing!
-Kenneth
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Questions
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller Sent: Tuesday, March 06, 2007 7:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Questions Dave Fullerton wrote on 3/6/07 9:33 AM: Polycom's 2.1.0 firmware has the new DST settings as the default. This is what they use for the SNTP element: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ The one thing I'm not sure about is the tcpIpApp.sntp.daylightSavings.start.date=8 line. According to the 2.1.0 admin guide that means the second week of the month but none of the guides before that mention this as a valid option. Thanks! One question I have... with this applied (and even with the original config I had before changing it to this), the start.dayOfWeek setting shows up as Monday on the web interface on the phone. Is the web interface goofed up, or should that be Sunday? -- I saw an article on the Polycom knowledgebase saying this is a bug in the web interface... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GTalk/Jabber passing audio in 1.4.1!
I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! - Ronald Lewis http://ronaldlewis.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nomination for Coolest App in 2007
Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I am totally floored with how cool it is! Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Long term voicemail archival and synchronisationbetween multiple storage locations?
I guess I would not consider this an advanced application. I have something that will do this will all conversations, not just voicemails. Voicemail should be trivial. Thanks, Steve Totaro _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, March 06, 2007 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Long term voicemail archival and synchronisationbetween multiple storage locations? Are any asterisk developers currently providing the following services on their asterisk either on-premise installations or hosted asterisk services? 1/ Hierarchical access to voicemail that can allow a local user to 'delete' a voicemail locally but also enable HR/IT Security to retrieve a voicemail for auditing/SEC/HIPAA requirements at a later date the same way emails are archived for long term access. 2/ In addition is anyone currently offering long term centralized storage for their customers in a telco grade data center. For example, I delete voicemail locally on my server but it is flagged to be sent to long term archives, my asterisk server once a day or routinely over the course of the day synchronizes with a centrally located database in a secure facility with multiple redundancies that stores these voicemails for later access via a hierarchical and logged web application. This service could be offered on a per minute/per gb basis for high profitability long term storage. If anyone is currently offering these functions or has developed these functions please contact me via phone or email. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph (P.S. I guess this is what I was asking for last week when I asked for advanced application development rather than coding ex-girlfriend routines) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System
J French wrote: We have a dinosaur Meridian system (~version 2) with 4 digital lines going to a Repartee Voicemail server. The Repartee got smoked by lightning two days ago and I'm itching to get Asterisk installed in its place. PRI is not an option since the system is so old that it doesn't even support PRI. I need to figure out how to connect the old Meridian to Asterisk otherwise. Any advice on getting Asterisk to work in its place is really appreciated. Do you have any analog ports, either FXO for FXS? Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] gtalk2voip and Asterisk
After upgrading to Asterisk 1.4.1 from 1.4.0 it just worked for me. There must have been a bug in 1.4.0. I have successfully connected to a Gmail and MSN instant message client. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mani Sridhar Sent: Saturday, 3 March 2007 8:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] gtalk2voip and Asterisk hi, i was able to get this working with google talk. i entered [EMAIL PROTECTED] using the gtalk2voip.com website's invite box, and as a result, saw a request from [EMAIL PROTECTED] to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten = 3501, 1, Dial(SIP/[EMAIL PROTECTED]) this allows any extension in my asterisk box to dial 3501, and [EMAIL PROTECTED] receives the call on the google talk client. the call is established, and voice quality is good. to allow a call from google talk client for [EMAIL PROTECTED], i opened a chat window to the buddy [EMAIL PROTECTED] and typed call [EMAIL PROTECTED], and this made extension ABC on my asterisk box start ringing. again, the call was established, and audio was ok. as far as asterisk is concerned, this is a SIP call. bottomline - it's a good alternative to using the native jabber/jingle library in asterisk 1.4 . in fact, i haven't been able to get asterisk to successfully set up a call to googletalk using the chan_gtalk module . i am inside a NAT-ed LAN, and audio works in one direction only for the asterisk (SIP) - gtalk call. anyone else got asterisk - googletalk using chan_gtalk working? Message: 10 Date: Fri, 02 Mar 2007 19:07:41 +0200 From: Cosmin Prund [EMAIL PROTECTED] Subject: Re: [asterisk-users] gtalktovoip and Asteirsk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I don't think it works. I tried calling my own yahoo messenger ID with no success: it rings a number of times and then it goes to some sort of voice mail. And I did invite the user they specified to my yahoo list, I also entered my yahoo id into the registration form on the site. I used a extensions.conf command like this for the try: exten = 641,1,Dial(SIP/[EMAIL PROTECTED]) (and yes, that's one of the yahoo ID I tryed with, and I don't think it exists! ) Klaverstyn, David C wrote: Has anyone managed to get gtalktovoip working at all? If so please explain. http://www.gtalk2voip.com/faq.shtml *2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ?* A: This is a major feature of our gateway and it is very easy. oGTalk: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oMSN: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oYahoo: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- _ Get Married in 2007. Join Shaadi.com http://www.shaadi.com/ptnr.php?ptnr=mhottag ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anybody having problems using sellvoip?
International calls (Germany) haven't completed since around 3/1. Domestic works. Is it just me? I'm getting 503 responses. Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Linksys PAP2 and Caller ID
Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to show the Caller number on the phone. There's a Caller ID Method: option on Regional settings, but I tested all options, and my CLIP phone never shows the Caller number... It should work fine. First, verify that you have for Line 1 (if phone connected to port 1): CID Serv: yes CIDCW Serv: yes and for User 1: CID Setting: yes CIDCW Setting: yes Of course, you must not answer until several seconds after the first ring completes. If you are using distinctive ring (or Default Ring is not 1), there are many subtleties that may be causing your trouble; try without it. If no luck, use a butt-set or similar to check whether CID modem tones are present after the first ring. If yes: Test the phone on a POTS line or another service to be sure CID is working ok. If so, try playing with ringing voltage, frequency, or waveform. If no: Use SIP Debug (or networking tools) to look at the SIP received by the PAP2; confirm that Asterisk is sending valid CID info. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 21
-- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson [EMAIL PROTECTED] Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Friends in the Asterisk community, One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I've constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I've been kind of avoiding during my trainings is voicemail. And there's where I've spent a lot of time recently... Life is strange. Instead of fixing the current voicemail, I decided to restart. Break up large apps into small building blocks, allowing Asterisk admins to use the rich dialplan script language or AEL to build a voicemail solution that fits the organization. I've named this minivoicemail, which for each addition becomes more of a bad choice of name for this project. Flexivoicemail could be better... :-) I've removed functionality like ODBC and IMAP support, something that can be reapplied later. I've also not replaced the hooks into other channels for voicemail notification, but that can be done too. I haven't started replacing voicemailmain(), since I've focused on the need of larger systems where one only supports e-mail notifications of voicemail with audio attached. What I currently have is: Applications - MinivmGreetPlay voicemail greetings (busy/unavailable/temporary) - MinivmRecordRecord voicemail message - MinivmNotifyNotify account owner of message (email, pager) - MinivmDeleteDelete message Functions - MINIVMACCOUNT() - Get properties of voicemail account CLI commands - minivm show settings - minivm reload - minivm show stats - minivm list accounts - minivm list templates New features: - I've added support for e-mail and pager templates in various languages. - All apps are usable without setting up a voicemail account for a user. Just run the app with an e-mail address as an argument. The branch is based on Asterisk 1.2 and can easily be downloaded from http://svn.digium.com/svn/asterisk/team/oej/minivoicemail I need testers, ideas for new applications and possibly coders that can help to complete this. To start - Checkout this branch, compile and install - Check the minivm.conf.sample for instructions - Read the top of the source code file for ideas, todo's and changes http://svn.digium.com/view/asterisk/team/oej/minivoicemail/apps/ app_minivm.c?view=markup (And if you want to encourage me further, paypal to [EMAIL PROTECTED], thanks) Thanks for your help building a more flexible voicemail system for Asterisk! Send bug reports, comments and ideas directly to me and I'll try to summarize. /Olle Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Back to back E1 - asterisk = toshiba pbx - Call droping issue
Hi Team, I have integrated asterisk with Toshiba analog PBX. NOw the live setup is going. Now I am facing call droping problem. It's happening ample time. 10-20 calls are droping every day. What could be the reason. I attached latest zapata.conf file for your information. This is being a huge issue. Highly appreciate your help on this regard. Thanks Regards, Vidura Senadeera. On 1/26/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear Marco, There is a huge problem i'm facing. My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i conected to the telco. other E1 port i'm using to cros-connection with toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx E1 not getting. d-channels are not getting up. what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12. notes - if i put, zap show channels in asterisk cli. its only showing the first 31 channels. but with ztcfg -vvv it showing al the channels. my configs are # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 RED # Suntel E1 connection == span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 # Span 2: WCT1/1 Digium Wildcard TE110P T1/E1 Card 1 # Legacy PBX E1 connection === span=2,2,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # Span 3: WCTDM/0 Wildcard TDM2400P Prototype Board 1 fxoks=63 fxoks=64 fxoks=65 fxoks=66 fxoks=67 fxoks=68 fxoks=69 fxoks=70 fxoks=71 fxoks=72 fxoks=73 fxoks=74 fxoks=75 fxoks=76 fxoks=77 fxoks=78 fxoks=79 fxoks=80 fxoks=81 fxoks=82 fxsks=83 fxsks=84 fxsks=85 fxsks=86 # Global data loadzone= us defaultzone = us Regards, vidura -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. zapata.conf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail question
Everything can be done with a certain amount of coding... :-) No, it's not possible in Asterisk today. Check the configuration templates to make life easier when configuring voicemail. It's documented in doc/configuration.txt in your asterisk source code directory, or here: http://svn.digium.com/view/asterisk/branches/1.4/doc/configuration.txt /Olle 6 mar 2007 kl. 00.42 skrev Hall, Eric M.: Group In voicemail.conf I would like to having the following setup per context not per-mailbox settings serveremail userscontext fromstring usedirectory emailbody pagerfromstring dialout sendvoicemail callback review operator volgain nextaftercmd forcename forcegreetings tempgreetwarn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
