[Asterisk-Users] IAX2 and FWD - Wrong context?
Title: IAX2 and FWD - Wrong context? Greets all, I'm setting up asterisk and trying to get IAX2 running for FWD. I followed the FWD IAX2 page verbatim but I get the following error May 14 08:09:31 WARNING[7569]: chan_iax2.c:5569 socket_read: Call rejected by 65.39.205.121: No such context/extension The rsa key is in, that's the only error I'm seeing when I try calling out with this. I haven't tried inbound yet. Any help would be appreciated. Thanks, Don Fanning Freelance Hacker - Producer of the 3 M's (Music, Movies and Microcode) Wherever you go, There you are. - Buckaroo Banzai ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDMoE emulates a T-1= Is there a product tosimulate a PRI trunk? (Robert Goodyear)
On Fri, 13 May 2005, jltaylor wrote: Does the TDMoE only allow one T1 per segment? You can add an index to have several TDMoE links and thus several virtual T1/E1 links between two computers. TMDoE is mostly used to provide an interconnect with a low latency over ethernet. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to beark Queue() and jump to voicemailMain
Hi to all
I have setup asterisk server setup with TDM400p and working very fine
using analog phones as well as four zap channels.
Now for incomming call i have setup a que like this
exten = _11.,1,Answer
exten = _11.,2,Macro(record-enable)
exten = _11.,3,Macro(vmail1)
...
[macro-vmail1]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Queue(dave|tH|||120)
exten = s,6,Wait,1
exten = s,7,Voicemail(b${MACRO_EXTEN:2})
exten = a,1,VoicemailMain(${MACRO_EXTEN:2})
exten = #,1,Hangup
Now all process works fine i can leave and check voice mail agiant
each of my number
But i have porblem in voiceMailMain()
i have to wait upto 120 sec to check voicmail by pressing *. i have
done all of my efferrts to solve it so that i can check my voice mail
on same extension without waiting 120 sec as i donot want ot shorten
queue time .
I will highly apperciate any help in this regard,
Cheers,
Mazhar
System Administrator
Nettechltd.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1-800 with FWD
And their 612/613 etc numbers should be dialed as 393612 etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick M. Gray, Jr. Sent: Friday, May 13, 2005 7:15 PM To: 'Juanjo Portela'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 1-800 with FWD Did you dial the 800 number correctly? You need to dial *1800XXX. I had this problem for a while and then checked out the docs on FWD's website. Any toll-free number seems to require a * before dialing. You can setup your dialing prefixes to add it automatically so it becomes transparent to users. Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juanjo Portela Sent: Friday, 13 May, 2005 19:07 To: Lista Asterisk Subject: [Asterisk-Users] 1-800 with FWD Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial(SIP/Phone4-e85b, SIP/[EMAIL PROTECTED]|90|Ttr) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 500 I'm terribly sorry, server error occured (1/SL) back from 69.90.155.70 -- SIP/fwd.pulver.com-f526 is circuit-busy == Everyone is busy/congested at this time Have you any idea? Thank you in advance, Juanjo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to connect two Asterisk servers
Hi Pls I want to know how to connect two Asterisk servers with sip,on the voip-info.org the iax details exist but the sip there is nothing about its details,pls any one can help. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 with FWD
And their 612/613 etc numbers should be dialed as 393612 etc. Where did you get that? Service numbers are dialed as they are published on http://www.freeworlddialup.com/advanced/service_numbers with no prefix other than one you may have arbitrarly added in your own dialplan. 393 (fwd on the dial) is often used in dialplans. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 with FWD
-- Executing Dial(SIP/Phone4-e85b, SIP/[EMAIL PROTECTED]|90|Ttr) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 500 I'm terribly sorry, server error occured (1/SL) back from 69.90.155.70 -- SIP/fwd.pulver.com-f526 is circuit-busy == Everyone is busy/congested at this time Here's what I get: -- Called [EMAIL PROTECTED] -- SIP/fwd.pulver.com-4634 is ringing -- SIP/fwd.pulver.com-4634 answered SIP/2002-0997 So your dial statement is correct, but what does your [fwd.pulver.com] sip peer entry look like? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax service (instead of tdm card)
There seem to be a lot of these companies popping up and going away again, each with their own limits and flaws. TrustFax for example, only allows you to fax North American numbers. I still have an account with j2.com and it works fine but has gotten too expensive ($15/mo). They do fax, conferencing and vmail by the way and offer numbers worldwide. I think today's best solution would be the big providers themselves offering the solution either for your existing DID or as a second fax-only number. Don't some of them (voicepulse, i connect here) do this now? I know I will want to replace j2 within the year and have been trying spandsp for months. It receives spam faxes 100% but a few customers faxes don't work with it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
Lt.Uhura RadarOriely Spock Tpol ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
what would you name them? Since we only have one box and it is a pbx, I call her Phoebe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 and FWD - Wrong context?
I'm setting up asterisk and trying to get IAX2 running for FWD. I followed the FWD IAX2 page verbatim but I get the following error May 14 08:09:31 WARNING[7569]: chan_iax2.c:5569 socket_read: Call rejected by 65.39.205.121: No such context/extension How about giving us a look at your dial exten ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax service (instead of tdm card)
On 13 May 2005, at 23:38, Terje Elde wrote: Hi all, Sorry if this is too far off-topic, it sounds potentially interesting to others though. I'll be brief. Rich Adamson wrote: I gave up (for now) trying to make spandsp work with the digium TDM card. Instead, I signed up with www.trustfax.com at a cost of $9.95 per year plus $.10/page. Since we only deal with an estimated 120 pages per year, the total cost of about $22/year seemed like a very reasonable alternative. (At least until we can find out why the TDM card does not function properly with spandsp.) I've just started playing with FAX, and I think I've got a solution that seems to work pretty well, others might want to try. Some background: We don't send or receive large numbers of faxes, and I run * on a slightly underpowered box (1Ghz Nemiah) so I was very wary of running spandsp and tiff etc on the * box, given that it has an E1 to manage and also does some transcoding to GSM and G729. As an experiment I plugged a spare port on a Sipura 2000 into the modem socket of my Apple G5, configured the sipura and * to only use alaw (the native codec of my E1 that goes into my * box), configured * to send a spare DID to that sipura port and told Macos X that it had a fax modem. So I now have a solution which seems to work well. The really cute thing is that the Apple presents the outbound fax as a unix postscript printer, so any computers in the office can print to it and send faxes! Inbound the faxes get converted to PDF and can be sent to email, printer or file, or any mix of the above. The only problem is night time faxes (from folks in other timezones). I put the G5 in sleep mode when I leave the office. In theory it should be possible to have the G5 wake up from sleep mode when the fax line rings, but it seems to be too slow for the sipura or something, the first attempt fails, but if the sender retries before the G5 goes back to sleep, then it works fine. Given that the imac minis are only a few hundred dollars, I thought this might be a solution that people would be interested in. Tim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to connect two Asterisk servers
Pls I want to know how to connect two Asterisk servers with sip,on the voip-info.org the iax details exist but the sip there is nothing about its details,pls any one can help. Its quite simple: Server2 (name obelix) should register at server1 (name asterix) 1. Enter in asterix: /etc/sip.conf [obelix] ;secret= username=obelix from_user=obelix type=friend context=default host=dynamic nat=no 2. in obelix: /etc/sip.conf enter the following in the section [general] register = [EMAIL PROTECTED] 3. To forward a call from obelix to asterix simply use the following: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,Ttr) Hope this helps Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What do you name yours
Wilson Pickett [EMAIL PROTECTED] writes: Since we only have one box and it is a pbx, I call her Phoebe How about Ernestine? A gracious hello!. :-) -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 Can't be unlocked
Are you trying to go right to 7.4? I had to install 6.3 first and then I could install 7.4. Others have had to upgrade in version increments. (ie from 3 to 4 to 5 to 6 to 7) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Friday, May 13, 2005 6:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 Can't be unlocked OK...now that I've been able to get in to the phones, I have (yet another) strange problem: I've configured the phone with a Static IP for testing. TFTP server settings are pointed at a known good TFTP server that other machines on the network are able to access and GET files from. When the phone boots up, it will appear on the network for about 5 pings worth of time, and then dissappear, only to reappear about 20 pings later. The phone's status message indicates a tftp server timeout. My tftp server's logs do not indicate any TFTP activity. The phones are Fw ver 3.1 MF.G2. (SCCP) I'm trying to convert them to SIP (obviously.) Any help will be greatly appreciated. John Mensel On Thursday 12 May 2005 13:06, John Mensel wrote: Tim, Thank you, that took care of the problem -- I'm much obliged. John On Thursday 12 May 2005 11:51, Timothy R. McKee wrote: Those are SCCP based phones. move the cursor to option 3, but do not press select. press **#, then press select. You should see the padlock icon with an unlocked appearance. press 32 and see if you have a YES option (alternate TFTP). If so press yes, then go to option 8 and edit the ip address. The phone sometimes locks itself in the middle and I have to start over. tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Thursday, May 12, 2005 12:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 Can't be unlocked Odd problem here--I just got a couple of Cisco 7960s from Ebay that are not functioning as expected.. These 7960s can't seem to be unlocked for manual configuration via any mechanism that I can find. If you go to settings, there is no option 9 (unlock). Available options stop at 4 (Status). **# has no effect. The Phones report that thier current firmware version is 3.1 MF.G2. When plugged into a known good DHCP/TFTP server, the phones will *sometimes* get a DHCP lease that is reflected in SettingsNetwork Configuration, but at no point will they grab new firmware via TFTP. DHCP server logs show the phones trying acquire a lease and then immediately requesting a new one. If anyone has encountered a similar situation, please advise. Thanks, John Mensel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax service (instead of tdm card)
Sorry if this is too far off-topic, it sounds potentially interesting to others though. I'll be brief. Rich Adamson wrote: I gave up (for now) trying to make spandsp work with the digium TDM card. Instead, I signed up with www.trustfax.com at a cost of $9.95 per year plus $.10/page. Since we only deal with an estimated 120 pages per year, the total cost of about $22/year seemed like a very reasonable alternative. (At least until we can find out why the TDM card does not function properly with spandsp.) I've just started playing with FAX, and I think I've got a solution that seems to work pretty well, others might want to try. Some background: We don't send or receive large numbers of faxes, and I run * on a slightly underpowered box (1Ghz Nemiah) so I was very wary of running spandsp and tiff etc on the * box, given that it has an E1 to manage and also does some transcoding to GSM and G729. As an experiment I plugged a spare port on a Sipura 2000 into the modem socket of my Apple G5, configured the sipura and * to only use alaw (the native codec of my E1 that goes into my * box), configured * to send a spare DID to that sipura port and told Macos X that it had a fax modem. So I now have a solution which seems to work well. The really cute thing is that the Apple presents the outbound fax as a unix postscript printer, so any computers in the office can print to it and send faxes! Inbound the faxes get converted to PDF and can be sent to email, printer or file, or any mix of the above. The only problem is night time faxes (from folks in other timezones). I put the G5 in sleep mode when I leave the office. In theory it should be possible to have the G5 wake up from sleep mode when the fax line rings, but it seems to be too slow for the sipura or something, the first attempt fails, but if the sender retries before the G5 goes back to sleep, then it works fine. Given that the imac minis are only a few hundred dollars, I thought this might be a solution that people would be interested in. The context for the original posting was ~$20/year (with the choice of several different service providers) is a reasonable alternative to using spandsp with the TDM analog card. E1/T1 users are not necessarily subjected to the same problems as the TDM analog card. The use of the TDM analog card implies a small (typically soho) one-to-four pstn line asterisk system where the quantity of faxes is fairly low, and seems to be a rather prevalent system for lots of US users. With such an external service provider, there is one less need for an analog pstn line dedicated to faxes (regardless of what is hanging on the end of the analog pstn line). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, 13 May 2005, Elmar Haneke wrote: Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. There are at least two anoying bugs: 1. The Busy-Applicatzion does not work, there seems to be no was to singnal Busy to the caller is no SIP-Phone is ready to answer the call. 2. Dial-Application does not really detect the reason for Failings. As an Example you should have a look at the LCR script available at Telefonsparbuch.de: The script trys to do some Fallback but it does not work with chan_capi. Thanks, for pointing out such issues. But can you please be more specific and give an example on how to reproduce it? For example, if you use an Point-to-Multipoint ISDN connection (not 'Anlagenanschluss'), then you won't get an immediate 'BUSY' on SIP Busy/Congestion. It's not possible to signal the caller 'Busy' or 'Reject', because there is a timeout on the ISDN-Bus for ANY OTHER device which may answer the call. Only on timeout, the Busy is signaled. So what type of connection and environment do you use? Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] installing linksys pap2 and welltech lp302
Hello list, I have spent the last couple of days installing a network of Linksys PAP2-NAs and some Welltech LP302s linked to an * box. I have found that they work great and have posted the configuration files on AstRecipes, so they can be shared. See http://www.oinko.net/astrecipes/index.php?n=84 for the PAP2 and http://www.oinko.net/astrecipes/index.php?n=83 for the LP302. If anybody else has experience with those phones and would like to add information or update any mistakes, please let me know. Thanks l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help Please Multiple Users for Broadvoice
I would like to be able to have multiple users (the wife and kids) to be able to access the Broadvoice account at the same. No complaining that way from them J. I seen someones configuration in the group here but now I cant find it (lost my glasses). If someone could post theirss or the shortcut that would be great. Thanks for your help. Dad shes on the phone again ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installed ztdummy, Asterisk doesnt work anymore
Michel Bachofen schrieb: Hi Since Im using the mISDN drivers and no zaptel stuff, I had to install ztdummy to get MeetMe to work. Well, that was the plan. Now, after getting the latest zaptel version over CVS (Im using Kernel 2.6), uncommenting all the modules except ztdummy in zaptel.sysconfig file and compiling this by make, make install and make linux26, I rebooted and recompiled Asterisk with make install again. Before all this, Asterisk worked perfectly fine, incoming and outgoing connections worked, everything did its job (except the MeetMe of course). But now I get the following errors when starting up Asterisk: [ Booting.Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring switchtype Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring pridialplan Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring overlapdial Apr 19 11:07:28 ERROR[6237]: chan_zap.c:9436 setup_zap: Unknown signalling method 'bri_cpe_ptmp' Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring overlapdial Apr 19 11:07:28 ERROR[6237]: chan_zap.c:9078 setup_zap: Signalling must be specified before any channels are. Apr 19 11:07:28 WARNING[6237]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 Apr 19 11:07:28 WARNING[6237]: loader.c:440 load_modules: Loading module chan_zap.so failed! Now, I havent done anything to the zaptel.conf yet, Im not sure if I even have to (didnt have that file before all this)? Looking forward to hearing anything in this matter :-( Michel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Michael, do you have compiled and installed libpri!? Jörg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pbx autodiscovery
Hello list, is there a way to configure a SIP phone to autodiscover its own PBX ona LAN? when running H.323, it is quite trivial to set up the Gatekeeper autodiscovery so that you can have all units working with dynamic DHCP addresses and you do not have to configure each unit by hand when you move them out of the lab and on to the client site. Is there any way to have anything similar with SIP, or any smart strategy to avoid reconfiguring every box by hand when moving to a different IP address? Thanks l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installed ztdummy, Asterisk doesnt work anymore
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jörg Steiner wrote: Michel Bachofen schrieb: Hi Since Im using the mISDN drivers and no zaptel stuff, I had to install ztdummy to get MeetMe to work. Well, that was the plan. Now, after getting the latest zaptel version over CVS (Im using Kernel 2.6), uncommenting all the modules except ztdummy in zaptel.sysconfig file and compiling this by make, make install and make linux26, I rebooted and recompiled Asterisk with make install again. Before all this, Asterisk worked perfectly fine, incoming and outgoing connections worked, everything did its job (except the MeetMe of course). But now I get the following errors when starting up Asterisk: [ Booting.Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring switchtype Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring pridialplan Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring overlapdial Apr 19 11:07:28 ERROR[6237]: chan_zap.c:9436 setup_zap: Unknown signalling method 'bri_cpe_ptmp' Apr 19 11:07:28 WARNING[6237]: chan_zap.c:9615 setup_zap: Ignoring overlapdial Apr 19 11:07:28 ERROR[6237]: chan_zap.c:9078 setup_zap: Signalling must be specified before any channels are. Apr 19 11:07:28 WARNING[6237]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 Apr 19 11:07:28 WARNING[6237]: loader.c:440 load_modules: Loading module chan_zap.so failed! Now, I havent done anything to the zaptel.conf yet, Im not sure if I even have to (didnt have that file before all this)? Looking forward to hearing anything in this matter :-( Michel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Michael, do you have compiled and installed libpri!? Jörg The build order should have been: make linux26 make install an you may have to make the file modifications for udev (see README.udev). - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQoYD0EtP/KMNOfRbAQJ+uwf8DY5E1Y614FaIK00Cd921kMG0LAB7NOQD SBPNFi+R5N1P2vygmfFRVZFcO81z0oTJNgQvvxf66U9OkAo0i9KeahOhn+Y4Z2TL QydbhWhBKZqTqKBVEcndulHl3BgtFZtJPx/xqous3zc22P4YaxcCVOyU7URO8AiR aU02zUcbQrYD+bS0eK6MBpmeX0faSIQZP9kzF03VaSUjbZJahabNwSwK7J2JBD21 y1QIQYe34ER0pNb3+40Z2CvLvVVWh71qYLalNc+Tvz/iXqROgWzL9oGxNb0ydTdG eyhvxrq1bj/HX4JWhq96nN0WGcrmLQxsEkXjyMg4EvvEnw3VY5Pf4Q== =fL2l -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - fax - spandsp
u said: I will try upgrading to a newer kernel (I'm using Fedora Core 2 with 2.6.8 kernel), turn off hyper-threading and see if it makes any difference... i said: wierd. Im running fc2 2.6.8 smp no problems. Could be timing slips on your PRI, happened to me until I looked hard at the PRI good luck let me know ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fix for chan_capi-0.3.5 using kernel = 2.6.11
Hi all, I found the fix for the problem of 'no sound on outgoing capi calls'. It is not an error of the kernel (capi or driver), it an error of chan_capi-0.3.5. A wrong file-descriptor is used for the pipe to asterisk. I don't know why this worked with other kernels, maybe kernel 2.6.11 got a 'fix' here. Armin Here is the patch to fix chan_capi-0.3.5: diff -ur chan_capi-0.3.5.orig/chan_capi.c chan_capi-0.3.5.new/chan_capi.c --- chan_capi-0.3.5.orig/chan_capi.c2004-08-13 12:07:28.0 +0200 +++ chan_capi-0.3.5.new/chan_capi.c 2005-05-14 15:47:36.164052000 +0200 @@ -687,7 +687,7 @@ p = malloc(sizeof(struct capi_pipe)); memset(p, 0, sizeof(struct capi_pipe)); p-fd = fds[1]; - c-fds[0] = fds[1]; + c-fds[0] = fds[0]; p-PLCI = -1; p-i = i; p-c = c; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 and FWD - Wrong context?
Here ya go, thanks!
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the
';')
; Note that this is different from the include command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
CONSOLE=Phone/Phone1
;IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=Zap/g2 ; Trunk interface
;TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
OUTBOUND=SIP/voipuser ; outbound calls route to www.voipuser.org
DON=SIP/10001 ; Don's Extension
ALAN=SIP/10002 ; Alan's Extension
EVERYONE=${DON}${ALAN} ; All Extensions
[default]
include = mainmenu
;*
;***FWD IAX Parameters
;*
;
FWDNUMBER=xx ; your calling number
FWDCIDNAME=GCSDP ; your caller id
FWDPASSWORD=xxx ; FWD password
FWDRINGS=10003 ; the phone to ring
FWDVMBOX=0 ; the VM box for this user
[trunkint]
;
; International long distance through trunk
;
exten = _00.,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _00.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten = _1NXXNXX,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _1NXXNXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _NXX,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _NXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten = _1800NXX,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _1800NXX,2,Congestion
exten = _1888NXX,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _1888NXX,2,Congestion
exten = _1877NXX,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _1877NXX,2,Congestion
exten = _1866NXX,1,Dial(${OUTBOUND}/${EXTEN},tT)
exten = _1866NXX,2,Congestion
[international]
;
; Master context for international long distance
;
;ignorepat = 0
;include = longdistance
;include = trunkint
[longdistance]
;
; Master context for long distance
;
;ignorepat = 0
;include = local
;include = trunkld
[outbound]
;
; Master context for local, toll-free, and iaxtel calls only
;
include = default
include = parkedcalls
include = trunklocal
include = trunktollfree
include = emergency
include = pobk_iax
include = fwd-out
include = nexthopproject
include = iaxtel-out
[inboundinternal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
include = outbound
include = internalextensions
include = voicemail
include = conference
include = sillystuff
include = starsixnine
include = parkedcalls
[fromiaxfwd]
exten = ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten = ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten = ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
[inbound-from-sip]
;purged to default context
include = default
[inbound-from-iax]
;purged to default context
include = default
;unless they dialed a 10001 extension, in which case
exten = _1000x,1,Goto(internalextensions,${EXTEN},1)
;transfer them to the internalextensions/extensionnumber priority 1
exten = _.,1,Goto(mainmenu,s,1)
exten = _.,2,Hangup
;redundancy
exten = i,1,Goto(mainmenu,s,1)
;invalid extension?
;ok set caller id to 7callerid
exten = i,1,SetCIDNum(7${CALLERIDNUM})
;then goto main menu
exten = i,2,Goto(mainmenu,s,1)
;make sure they hangup
exten = i,3,Hangup
;called anything?
;ok set caller id to 7callerid
exten = _,1,SetCIDNum(7${CALLERIDNUM})
;then goto main menu
exten = _,2,Goto(mainmenu,s,1)
;make sure they hangup
exten = _,3,Hangup
[incoming_voipuser]
exten = 08449864758,1,NoOp(---${CALLERID} calling on VoIPUser (${EXTEN})
---)
exten = 08449864758,2,Dial(SIP/10003,20)
exten = 08449864758,3,Answer
exten = 08449864758,4,Wait,1
exten = 08449864758,5,Voicemail(u1)
exten = 08449864758,6,HangUp
[internalextensions]
;internal extensions obviously, call us using the macro stdexten
exten =
RE: [Asterisk-Users] Help Please Multiple Users for Broadvoice
http://geekgazette.com has a "how to" for Broadvoice -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mr. barkerSent: Saturday, May 14, 2005 6:03 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Help Please Multiple Users for Broadvoice I would like to be able to have multiple users (the wife and kids) to be able to access the Broadvoice account at the same. No complaining that way from them J. I seen someones configuration in the group here but now I cant find it (lost my glasses). If someone could post theirss or the shortcut that would be great. Thanks for your help. Dad shes on the phone again ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1-800 with FWD
http://www.voip-info.org/wiki-AsteriskAtHomeFWD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Saturday, May 14, 2005 1:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1-800 with FWD And their 612/613 etc numbers should be dialed as 393612 etc. Where did you get that? Service numbers are dialed as they are published on http://www.freeworlddialup.com/advanced/service_numbers with no prefix other than one you may have arbitrarly added in your own dialplan. 393 (fwd on the dial) is often used in dialplans. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring a call, IAX2-SIP, DTMF/RFC2833 doesn't work?
We are using Asterisk 1.0.7. We have this scenario: IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user. The dial command is simple, looks like this: exten = 300,1,Dial(SIP/300) Extension 300 is a legacy PBX device operated by touchtones. The user (coming in over IAX2) is trying to drive this PBX using touchtones. But the trouble is, by the time the touchtones go out to the SIP extension, it's being sent as audio, not as RFC2833. An ethereal trace confirms this. DTMF is arriving to Asterisk (via IAX2) as OOB data, but no RFC2833 is going back out to the SIP device. The SIP device is configured to use RFC2833 wherever possible. This doesn't work because the DTMF is arriving really choppy-sounding, and the PBX doesn't recognize it. Am I doing something wrong? Help would be appreciated! P.S. When we go IAX2--Asterisk--IAX2, DTMF OOB is preserved correctly. This is just when we go IAX2--Asterisk--SIP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] French SIP or IAX phones
Nous avons des telephones IP qui support IAX2 et SIP en version francaise. mail: [EMAIL PROTECTED] Cordialement Jorge Mendoza [EMAIL PROTECTED] : PolycomJorge MendozaMartin Roy wrote: Is there any SIP or IAX phones that can be configure in french instead of english. I tested Cisco 7960 phones but I can't change the language it's only available in english with the SIP firmware. I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to french. Thanks Martin Roy___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersDo You Yahoo!? 150MP3 1G1000___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
chan_capi 0.3.5 lacks proper support for passing on isdn cause codes to Asterisk. This is already fixed in my development version and will be in 0.4.0. :) best regards Klaus Am Samstag, den 14.05.2005, 14:50 +0200 schrieb Armin Schindler: On Fri, 13 May 2005, Elmar Haneke wrote: Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. There are at least two anoying bugs: 1. The Busy-Applicatzion does not work, there seems to be no was to singnal Busy to the caller is no SIP-Phone is ready to answer the call. 2. Dial-Application does not really detect the reason for Failings. As an Example you should have a look at the LCR script available at Telefonsparbuch.de: The script trys to do some Fallback but it does not work with chan_capi. Thanks, for pointing out such issues. But can you please be more specific and give an example on how to reproduce it? For example, if you use an Point-to-Multipoint ISDN connection (not 'Anlagenanschluss'), then you won't get an immediate 'BUSY' on SIP Busy/Congestion. It's not possible to signal the caller 'Busy' or 'Reject', because there is a timeout on the ISDN-Bus for ANY OTHER device which may answer the call. Only on timeout, the Busy is signaled. So what type of connection and environment do you use? Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 with FWD
http://www.voip-info.org/wiki-AsteriskAtHomeFWD If anyone is using @home and needs help here, they'd best say that right away. I have no idea what @hole does but 393 is an added dial prefix that may differ in any install. In fact, one could, if using FWD to be the default 800 number service, just dial 1800 and have the dialplan add the *. 393 is a convention, it is not mandatory, unless @hole has imposed it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Make error trying to add app_dicate to Asterisk stable 1.0.7
I'm trying to add app_dictate to Asterisk stable 1.0.7 and get a make error 255. Can anyone suggest what might be wrong? Here's the output: [EMAIL PROTECTED] app_dictate]# make autoload CC=gcc /usr/src/asterisk/contrib/scripts/astxs app_dictate.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\1.0.7\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run/asterisk\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c app_dictate.c -o app_dictate.o app_dictate.c: In function `dictate_exec': app_dictate.c:66: variable `flags' has initializer but incomplete type app_dictate.c:66: warning: excess elements in struct initializer app_dictate.c:66: warning: (near initialization for `flags') app_dictate.c:66: storage size of `flags' isn't known app_dictate.c:84: warning: implicit declaration of function `ast_strlen_zero' app_dictate.c:85: warning: implicit declaration of function `ast_separate_app_args' app_dictate.c:132: warning: implicit declaration of function `ast_queue_frame' app_dictate.c:66: warning: unused variable `flags' make: *** [app_dictate.so] Error 255 [EMAIL PROTECTED] app_dictate]# Thanks, Malcolm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice outage times?
Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, Id like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Building OPENH.323 ERROR HELP PLEASE
Hello, I try to build the code, I follow all the instruction from http://www.openh323.org , i still get all this errors, can someone Help me please. Configuration: Console - Win32 Release Configuring Build Options This program cannot be run in DOS mode. Error executing c:\winnt\system32\cmd.exe. asnparser.exe - 1 error(s), 0 warning(s) Configuration: GUI - Win32 Release Compiling... pwlib.cxx ..\..\..\include\ptlib.h(138) : fatal error C1083: Cannot open include file: 'ptbuildopts.h': No such file or directory Error executing cl.exe. pwtest.exe - 1 error(s), 0 warning(s) Configuration: GUI - Win32 Debug Compiling... pwlib.cxx ..\..\..\include\ptlib.h(138) : fatal error C1083: Cannot open include file: 'ptbuildopts.h': No such file or directory Error executing cl.exe. pwtest.exe - 1 error(s), 0 warning(s) Thank. Franz [EMAIL PROTECTED] Tel. 011-504-2214062 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice outage times?
As far as I noticed, it's mostly random and seems to depend more on the origin/destination of the call rather than time of day. But that's not the point -- you shouldn't have to tweak your dialplan because a service only works sometimes. That's just isn't good enough. --Luki On 5/14/05, Tim Connolly [EMAIL PROTECTED] wrote: Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, I'd like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PBX replacement
Yes, a single T1 that would now go from the bell into the new Asterisk system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, May 13, 2005 11:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PBX replacement I misread your original post...I thought you were talking about an Avaya DIALER and not an Avaya PBX. The only concern with the CSU is the distance the cabling with the signal will have to travel. Avaya is big on requiring CSUs for their equipment, but they are not necessary for 0-133 ft distances. The only thing I see is that you may have to change some of the pin outs for the RJ45 ends. Is this a single T1? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, May 13, 2005 7:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PBX replacement The actual setup is like this: T1 from bell comes into a CSU, from the CSU goes into Avaya PBX for inbound and outbound calls. My question/concern is if I still need that CSU. Need to replace Avaya with Asterisk PBX using a Digium t1 card. I've seen some responses stating that the Digium T1 card has a built-in CSU so I won't need an external one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, May 13, 2005 10:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PBX replacement Is the T1 coming into the Avaya dialer for outbound or inbound calls? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, May 13, 2005 6:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PBX replacement No, this is a T1 from the bell company -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, May 13, 2005 9:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PBX replacement Are you using the cards with an Avaya dialer? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Loftis Sent: Friday, May 13, 2005 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; 'BJ Weschke' Subject: RE: [Asterisk-Users] PBX replacement --On May 13, 2005 8:45:12 PM -0300 Leandro Tenorio [EMAIL PROTECTED] wrote: Sure??? I think that most probably he still needs the CSU. Absolutely certain he doesn't. We're using the cards here. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.9 - Release Date: 5/12/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.9 - Release Date: 5/12/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.9 - Release Date: 5/12/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Broadvoice outage times?
On Sat, 2005-05-14 at 11:50 -0500, Tim Connolly wrote: Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, Id like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether I have not but hearsay from someone who spoke to an insider at BV indicated that everything should be fixed by wednesday. I have noticed (but I may be paying more attention during those times) that they seem to be doing more reboots and whatnot at 6am GMT/10pm PST. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice outage times?
I don't mind so much that calls fail occasionally, but the fact that Broadvoice will let the failed call ring for 15 or 20 seconds, answer the call, play a cute little call can't be completed message, then hangup...really frustrated me. Why can't they send back a busy or congestion signal like every other telco in the world so I can try to reroute the call on another trunk. Right now, I never see failed attempts because something is answering them! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luki Sent: Saturday, May 14, 2005 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice outage times? As far as I noticed, it's mostly random and seems to depend more on the origin/destination of the call rather than time of day. But that's not the point -- you shouldn't have to tweak your dialplan because a service only works sometimes. That's just isn't good enough. --Luki On 5/14/05, Tim Connolly [EMAIL PROTECTED] wrote: Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, I'd like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DynExtenDB
DynExtenDB is any one using it? does it work? see http://andreasotto.net/asterisk/ last updated 2002-12-15 No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.9 - Release Date: 5/12/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting 20+ asterisk servers together
On Mon, 9 May 2005, Vikram Rangnekar wrote: [Deleted] Can dundi or the switch statement help me get out of this mess ? Dundi will make this trivial. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting 20+ asterisk servers together
On Mon, 9 May 2005, David Choo wrote: Actually, this is whats facing me right now. I think Dundi will resolve the problem, but I've never really placed it to the test. Anyone tested Dundi? Best Regards, I run it in production on CVS-Stable and it works without a problem. Our usage of it is to have multiple gateway servers that can dynamically respond to outages without human intervention. I.E. a customer's dial-plan will be setup to return multiple routes and it will try them in order of preference. If one fails, the next available route will be tried and so on. Here is a snippet of some information from one of the working drafts that N2Net is putting together on using Dundi as the core of a Fault-Tolerant network. It is incomplete, and has some stuff that is a bit wrong, but it is a starting point. Much of the really intelligent sounding stuff is ripped right from the Dundi RFC... The two gateways will be reponsible for publishing weighted Dundi routes to their PSTN (PRI) and VoIP Termination carriers. They will also be responsible for recieving inbound PSTN and VoIP traffic and routing the calls back to the customer. DUNDi is designed to facilitate the sharing of resources that can be used to terminate phone numbers by using a peer to peer system, requiring no centralized controlling authority, no single point of failure, and no enforced heirarchy. In this way, systems sharing a dialplan across an enterprise or across the globe can be assembled in an ad-hoc manor, while retaining confidence in the accuracy of the routes that are supplied and the security of both the queries and the answers within the trust group. Route Weights - A Weight is a value indicating the relative cost or indirection of a particular published number. A lower weight represents a route which is more direct to the intended location. The lowest weight value is 0. The generally accepted route weights for the Dundi Peering network are 0,100 and 400. The HAC will publish route weights of 0-99 to internal, private peers, and 100-400 for external Dundi-E.164 and Dundi-Test networks. This will allow routes to be prioritized, such that PSTN will always be used where possible but VoIP routes will be available if PSTN access is not. This will also allow the HAC to preferentially route International traffic using a Least Cost Routing method, preferring VoIP routes, or a particular PRI span if neccessary. A weight of 0-24 indicates a PSTN route to a dedicated carrier. A weight of 25-49 indicates a failover PSTN route to a carrier. A weight of 50-74 indicates a Primary VoIP route. A weight of 75-99 indicates a Seconddary VoIP route. A weight of 100+ indicates a least-preferred route to a private or public Dundi connected network, such as Dundi-E.164, Dundi-Test or FWD-out. For what it's worth.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot create a personalized unavailable message
When I try to create a personalized unavailable message the VoicemailMain application says please record your message beeps and then goes straight to if you want to accept... I checked the output of /var/log/asterisk/messages and see the following: WARNING: Unable to open file /var/lib/asterisk/sounds/voicemail/default/4035/unavail.WAV: No such file or directory. and sure enough when I check the directory tree, the directory /var/lib/asterisk/sounds/voicemail doesn't exist Is there something I forgot to configure? I would think that these directories would be created automatically. I'm running Asterisk version CVS-v1-0-02/17/05-17:34:40 TIA, Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser and asterisk
Hello, I have noticed when ser forwards to asterisk, the last registered host from ser is always the subsequent callee whichever client dials. i.e. 4561 registers 4562 registers 4563 registers 4562 calls 4561. Asterisk shows 4563 dialing 4561. I am forwarding registrations and invites to asterisk. Is this correct? Many thanks, Spencer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom configuration
The new SIP firmware improves this. See the release notes for firmware 1.5.0. And I just posted the interesting bits to the list a day or two ago. Jerry wrote: You can use one appearance to register which gives you 2 calls with call waiting, then use the others for speed dialing. Or have multiple registrations and use your dial plan to hunt between them. I hear the new 1.0.5 may allow this to happen easier but have not tried yet. On May 13, 2005, at 8:57 PM, Chris Mason wrote: How do you configure your Polycom phones? Is it enough to configure one line appearance? Or is there a way to configure a roll over? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
Wilson Pickett wrote: what would you name them? Since we only have one box and it is a pbx, I call her Phoebe I used to name servers after women I knew. Michelle thought it was cute that I named one after her but then found out I named another one after a young lady she highly disliked. She asked Why did you name a machine after that ugly bitch?. I told her Because it's always going down. She bought me dinner for that one. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
HelloOn 15/05/2005, at 12:00 AM, Colin Anderson wrote:wierd. Im running fc2 2.6.8 smp no problems. Could be timing slips on your PRI, happened to me until I looked hard at the PRI So you turned off Hyperthreading on your linux box or not??What could I do to check if my PRI has timing slips? I have to say that my knowledge in this area is close to zero. The extent of my knowledge was to connect a TE110P card, configure it and run it. That's itJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with ShoreTel 210 (MGCP)
Typically what I have seen with wrong ports, is that * or the Gateway doesn't see the MGCP message at all. From your debug it looks like * is getting the MGCP message, it is just having trouble finding the gateway. I'm not so sure this will fix your problem, but I would try this. Have * mgcp.conf set to port 2427, then make sure your phone is listening and sending on port 2427, you may have to put this in your phone IP.IP.IP.IP:2427 if there is no port definition. Also, change the endpoint in your phone to be aaln/1 for line one and aaln/2 for line 2 and so on. There should be a way to change that; aaln/1 and phone/1 are typically used for endpoint definitions, but I see that most manufactures use aaln/x I would try the 2 above recommendations and then see if that helps clear up the problem. Duane Cox - Original Message - From: Ben Dugdale [EMAIL PROTECTED] To: Duane Cox [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 13, 2005 11:03 PM Subject: Re: [Asterisk-Users] Asterisk with ShoreTel 210 (MGCP) Duane Cox wrote: can you post your mgcp.conf file. Gladly, but I should point out that I brought the phone home, so the network numbers differ from those I stated before. My * server = 192.168.0.5 Phone Settings IP = 192.168.1.137 (routed subnet, not NAT. SIP and AIX work) MGC = 192.168.0.5 Here's everything that isn't a comment in mgcp.conf grep -v '^;' mgcp.conf [general] port = 2727 bindaddr = 0.0.0.0 [192.168.1.137] accountcode = 1000 ; record this in cdr as account identification for billing amaflags= billing ; record this in cdr as flagged for 'billing', 'documentation', or 'omit' context = local host= 192.168.1.137 wcardep = aaln/*; enables wildcard endpoint and sets it to 'aaln/*' another common format is '*' callerid= Duane Cox 123 ; now lets setup line 1 using per endpoint configuration... callwaiting = no callreturn = yes cancallforward = yes canreinvite = no transfer= no dtmfmode= inband line = aaln/1 ; now lets save this config to line1 aka aaln/1 And the current error messages: Asterisk Ready. *CLI May 13 20:48:06 NOTICE[21844]: chan_mgcp.c:1644 find_subchannel_and_lock: Gateway '192.168.1.137' (and thus its endpoint 'SHOR_001049007E83') does not exist ngrep host 192.168.1.137 interface: eth0 (192.168.0.0/255.255.255.0) filter: ip and ( host 192.168.1.137 ) # U 192.168.1.137:2427 - 192.168.0.5:2727 RSIP 2162 [EMAIL PROTECTED] MGCP 1.0.RM: restart.X-ShoreModel: S1. # I have made no corresponding entries in extensions.conf yet. Also, I noticed that the default port setting seems to be 2727, and that's what the phone seems to be talking to, but the mgcp.conf example and your config indicate 2427. Is that significant? Thanks, From the debug output it looks like * can not find the gateway in the mgcp.conf (* goes on to tell you it can not match the endpoint, because it first has to find the gateway device...) - Original Message - From: Ben Dugdale [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 12, 2005 6:51 PM Subject: Re: [Asterisk-Users] Asterisk with ShoreTel 210 (MGCP) Duane Cox wrote: Yes * can work with MGCP phones directly. You have a configuration issue. Glad to hear it! a typical mgcp.conf might be: [general] port= 2427 bindaddr= 0.0.0.0 [10.21.4.2] accountcode = 1123 amaflags= billing context = main host= 10.21.4.2 wcardep = aaln/* callerid= YOUR NAME 1231231234 callwaiting = no callreturn = yes cancallforward = yes canreinvite = no threewaycalling = no transfer= no dtmfmode= none line = aaln/1 Where does a person find a list of the mgcp.conf options and meanings? ( I've tried 'man mgcp' 'man mgcp.conf' and looked for info in the doc directory of the * source (I did make documentation at install) )? turn on MGCP debug mgcp debug and see what messages are going to and fro. I'm now using Asterisk CVS-HEAD-05/12/05-16:10:03 Here is what I see at the console: MGCP Debugging Enabled *CLI MGCP read: RSIP 11630 [EMAIL PROTECTED] MGCP 1.0 RM: restart X-ShoreModel: S1 from 192.168.90.209:2427 Verb: 'RSIP', Identifier: '11630', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines May 12 16:31:56 NOTICE[28300]: chan_mgcp.c:1644 find_subchannel_and_lock: Gateway '192.168.90.209' (and thus its endpoint 'SHOR_001049007E83') does not exist MGCP read: RSIP 11630 [EMAIL PROTECTED] MGCP 1.0 RM: restart X-ShoreModel: S1 Here is what I see with ngrep port 2727 interface: eth0 (192.168.90.0/255.255.255.0) filter: ip and ( port 2727 ) # U 192.168.90.209:2427 - 192.168.90.6:2727 RSIP 11625 [EMAIL PROTECTED] MGCP 1.0.RM: restart.X-ShoreModel: S1. I've changed mgcp.conf to pretty much
[Asterisk-Users] Asterisk Guru help needed for DISA troubles
I have a setup which allows users to access my asterisk box via FWD. That is, a user in say, France can call into a local access number for FWD, then hit number 7 which dumps them into a DISA request for a password, which then dumps them into my internal extension so they can dial out through a VOIP provider. Everything works fine until they enter their tones for the number they are calling -- DISA does not respond, just sits there twiddling its thumbs. The strange thing is that it works fine when a caller direct dials into my Asterisk box then they can access the internal extension and dail outward. So, I'm presuming this has something to do with codecs being passed from the FWD server(s). For FWD I have (in addition to other settings): disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc dtmfmode=inband and for the VOIP provider for calling out I have disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc dtmfmode=inband dtmfmode=inband Why would Asterisk understand the tone for transfering to the DISA extension, but stall with any subsequent tones, yet work 100% for a direct dial inward call? I'm perplexed. Thanks for any insights a guru could provide. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
Jean-Yves Avenard wrote: Hello On 15/05/2005, at 12:00 AM, Colin Anderson wrote: wierd. Im running fc2 2.6.8 smp no problems. Could be timing slips on your PRI, happened to me until I looked hard at the PRI So you turned off Hyperthreading on your linux box or not?? What could I do to check if my PRI has timing slips? I have to say that my knowledge in this area is close to zero. The extent of my knowledge was to connect a TE110P card, configure it and run it. That's it Jean-Yves In most cases the slips on T1s and E1s are because people have carefully configured their system to cause them. Probably because there is no clear explanation around of the right way to do things. Their careful configuration generally consists of ensuring they have the following lines in their zaptel.conf file (E1 example): span=1,0,0,cas,hdb3,crc4 span=2,0,0,cas,hdb3,crc4 span=3,0,0,cas,hdb3,crc4 span=4,0,0,cas,hdb3,crc4 This ensures they do not listen to any external source's opinion about clock speeds. As they centre of the universe, their Asterisk box expects the 4 systems connected to their 4 E1s to fall in line with the clock speed of the Asterisk box. (here's a clue: a PSTN switch contains an atomic clock as its clock source, and believes everyone else's clock it too flawed to listen to). Other people carefully configure their zaptel.conf file to contain lines like (again, an E1 example): span=1,1,0,cas,hdb3,crc4 span=2,2,0,cas,hdb3,crc4 span=3,3,0,cas,hdb3,crc4 span=4,4,0,cas,hdb3,crc4 and then attach spans 1, 2, and 3 to a PBX and span 4 to the PSTN. So, their primary source of clock is some grotty PBX, as it is listed as the number 1 option for obtaining a clock. They expect the PSTN to fall in line with the PBX's clock. Well, if the PBX has actually locked directly to the PSTN by another E1, that might work out. However, in most cases the Asterisk box is playing piggy in the middle, and the PBX should be syncing to the clock it gets from the Asterisk box. The right thing to do is to sync to the PSTN. The E1s connected to the PSTN should be listed as the lowest numbered clock sources, starting from 1. Places you never want to sync to should be set to zero. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Guru help needed for DISA troubles
On May 14, 2005 11:58 pm, Jeffrey Starin wrote: disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc dtmfmode=inband It doesn't take a guru. :-) you can't use inband signaling if you are allowing the possibility of compressed codecs such as gsm and ilbc. Use RFC2833. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Guru help needed for DISA troubles
Thanks for replying Andrew. I removed the gsm and ilbc codecs and changed dtmfmode to rfc2833 but still no luck. Same issue as before. DISA just sits there and doesn't do anything. Works fine on direct dial in calls, just when stuff is passed through FWD or another server that DISA doesn't respond to the tones. Any other suggestions? B.. Andrew Kohlsmith wrote: On May 14, 2005 11:58 pm, Jeffrey Starin wrote: disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc dtmfmode=inband It doesn't take a guru. :-) you can't use inband signaling if you are allowing the possibility of compressed codecs such as gsm and ilbc. Use RFC2833. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
