RE : [asterisk-users] CallerID not detected by TDM22B
Hello Aslay, In some country, this feature is a paid option from the TELCO side. In France the analog lines have not this feature enabled in standard, only the digital lines . Are you sure that it's actualy available in your case ? Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de aslay-pinwee Envoyé : vendredi 18 mai 2007 14:23 À : asterisk-users@lists.digium.com Objet : [asterisk-users] CallerID not detected by TDM22B Hi, I am using asterisk 1.4 with Digium TDM22B card. My system is running well except CALLERID. I have tried all options for cidsignalling, cidstart but no luck. Btw, I am living in Malaysia. From google web site, I found some one having the same problem with new version of asterisk but not in old versions. I do not want to try the old versions of asterisk. I really appreciate if someone can help me to solve the problem Regards ASLAY ___ --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 is not showing the correctIncomming CallerID
Hi Farook and the list, You have may be forgotten to input that in the misdn.conf file : nationalprefix=0 internationalprefix=00 dialplan=0 localdialplan=0 cpndialplan=0 Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed Envoyé : mercredi 16 mai 2007 06:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk is not showing the correctIncomming CallerID I forgot to give the asterisk logs pbx*CLI -- Executing Set(mISDN/2-2, FROM_DID=3722) in new stack -- Executing Gosub(mISDN/2-2, app-blacklist-check|s|1) in new stack -- Executing LookupBlacklist(mISDN/2-2, ) in new stack -- Executing GotoIf(mISDN/2-2, 0?blacklisted) in new stack -- Executing Return(mISDN/2-2, ) in new stack -- Executing Goto(mISDN/2-2, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(mISDN/2-2, user-callerid|) in new stack -- Executing NoOp(mISDN/2-2, user-callerid: 1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 0?report) in new stack -- Executing GotoIf(mISDN/2-2, 0?start) in new stack -- Executing Set(mISDN/2-2, REALCALLERIDNUM=1416222888) in new stack -- Executing NoOp(mISDN/2-2, REALCALLERIDNUM is 1416222888) in new stack -- Executing Set(mISDN/2-2, AMPUSER=) in new stack -- Executing Set(mISDN/2-2, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(mISDN/2-2, 1?report) in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp(mISDN/2-2, TTL: ARG1: ) in new stack -- Executing GotoIf(mISDN/2-2, 0?continue) in new stack -- Executing Set(mISDN/2-2, _TTL=64) in new stack -- Executing GotoIf(mISDN/2-2, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(mISDN/2-2, Using CallerID 1416222888) in new stack -- Executing Set(mISDN/2-2, modifiedcallerid=1416222888) in new stack -- Executing Set(mISDN/2-2, CALLERID(number)=1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 1?skipdb) in new stack -- Goto (ext-group,1,4) -- Executing Set(mISDN/2-2, __NODEST=) in new stack -- Executing Set(mISDN/2-2, __BLKVM_OVERRIDE=BLKVM/1/mISDN/2-2) in new stack -- Executing Set(mISDN/2-2, __BLKVM_BASE=1) in new stack -- Executing Set(mISDN/2-2, DB(BLKVM/1/mISDN/2-2)=TRUE) in new stack -- Executing Set(mISDN/2-2, RRNODEST=) in new stack -- Executing Set(mISDN/2-2, __NODEST=1) in new stack -- Executing GotoIf(mISDN/2-2, 1?REPCID) in new stack -- Goto (ext-group,1,14) -- Executing NoOp(mISDN/2-2, CALLERID(name) is ) in new stack -- Executing Set(mISDN/2-2, RecordMethod=Group) in new stack -- Executing Macro(mISDN/2-2, record-enable|903-909|Group) in new stack -- Executing GotoIf(mISDN/2-2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(mISDN/2-2, recordingcheck|20070516-140757|1179288477.1037) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(mISDN/2-2, No recording needed) in new stack -- Executing Set(mISDN/2-2, RingGroupMethod=hunt) in new stack -- Executing Macro(mISDN/2-2, dial|10||903-909) in new stack -- Executing DeadAGI(mISDN/2-2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '1416222888' dialparties.agi: Methodology of ring is 'hunt' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 903 to extension map -- dialparties.agi: Added extension 909 to extension map -- dialparties.agi: Extension 903 cf is disabled -- dialparties.agi: Extension 909 cf is disabled -- dialparties.agi: Extension 903 do not disturb is disabled -- dialparties.agi: Extension 909 do not disturb is disabled dialparties.agi: extnum: 903 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: extnum: 909 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: NODEST: 1 adding M(auto-blkvm) to dialopts: M(auto-blkvm) -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp(mISDN/2-2, Returned from dialparties with hunt groups to dial ) in new stack -- Executing Set(mISDN/2-2, HuntLoop=0) in new stack -- Executing GotoIf(mISDN/2-2, 1?30 ) in new stack -- Goto (macro-dial,s,30)
RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
Hi Gavin, A second Asterisk server replacing the provider (best way), or doing a loop between two different ISDN ports on a same card (worst way) must help you. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gavin Henry Envoyé : mercredi 9 mai 2007 09:40 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --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] Audio going blank for a few seconds and then comesback. What could be the reason?
Hi Zeeshan, Ethernet Network (or Switch) congestion ? QoS not realy effective ? Too high CPU load in Asterisk the server ? Who knows... You must check during a default. Good kuck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Zeeshan Zakaria Envoyé : mercredi 9 mai 2007 12:02 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Audio going blank for a few seconds and then comesback. What could be the reason? Hi, Everything was working fine on this 10 phone office, but for last few weeks they are complaining that audio goes blank for a few seconds during the conversation, and then comes back on. It goes blank for both parties. What are the possible causes for this to happen? -- Zeeshan A Zakaria ___ --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] Wildcard TE410P problem
Hi Alexander and the list, Have you well checked your E1 cable ? Sometime, you must use a crossed E1 cable (not an Ethernet one)... Check also without the crc check. How is your zapata.conf file ? Have you checked with a loop (crossed E1 cable) between two spans (one in TE the second in NT, of course) ? Best Regards, Francois BERGERET, France. ___ --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] Wildcard TE410P problem
Autocorrection mode : pri_cpe / pri_net rather than TE / NT ;-) -Message d'origine- De : Francois BERGERET [mailto:[EMAIL PROTECTED] De la part de '[EMAIL PROTECTED]' Envoyé : jeudi 3 mai 2007 21:03 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE : [asterisk-users] Wildcard TE410P problem Hi Alexander and the list, Have you well checked your E1 cable ? Sometime, you must use a crossed E1 cable (not an Ethernet one)... Check also without the crc check. How is your zapata.conf file ? Have you checked with a loop (crossed E1 cable) between two spans (one in TE the second in NT, of course) ? Best Regards, Francois BERGERET, France. ___ --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 do I do this in Asterisk?
Hi Christian,
Increase a variable in the menu loop, or exactly in the t and i
extensions like this :
exten = s,1,Wait(3)
exten = s,n,Answer()
exten = s,n,Set(LoopStep=1)
exten = s,n,Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Wait(1)
exten = s,n(menurestart),Background(your_announce)
exten = s,n,WaitExten(5)
exten = 1,1,GoTo(your_menu_context,1,1)
exten = 2,1,GoTo(your_menu_context,2,1)
exten = 3,1,GoTo(your_menu_context,3,1)
exten = t,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = i,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = h,1,NoOp(the caller has hung up)
I hope that can help and to have not introduced mistakes ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Christian
Envoyé : mardi 1 mai 2007 18:18
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] How do I do this in Asterisk?
Hi all,
I have created a menu from which the caller can select several options such
as being transfered to our phones and my mobile phone, meetme, etc. If the
caller press an invalid option i have set it to play a message like invalid
choice please try again. If the caller make three invalid choices i want the
call to be disconnected. what is the best way of doing that?
And finally i have set up an extention to which it is possible to record a
message but i then want to be able to specify what number the message should
be plaied for after recording is finished. Many thanks for all your help,
Christian
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RE : [asterisk-users] DISABLE 9?
Hi everybody ! I never use any prefix number to dial out. I prefer to do like any standard residential subscriber, not to force somebody to think : Oh no ! I have forgottent to input the 9 - or 0 - before to dial out !. Directly inputing the real number is more natural. Adding a prefix is an old way to go inherited from analog PABX integrators ;-) If a customer want that, ok, do it, to avoid to have to change his habits. If you have no obligation to do that, forget it ! Think a good dialplan instead of that... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de JNA Envoyé : dimanche 15 avril 2007 11:49 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] DISABLE 9? Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? ___ --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 : RE : [asterisk-users] DISABLE 9?
Hello again, They are many Asterisk servers outside of the US that use a different national plan... Here, in France, we are using _0Z for fixed national telephones lines, including _06 for national mobiles, _08 special (often higher) price calls, _00Z. For international calls, and few _XX and _ as specific services as Police, Firemen, our historical TELCO, some data only destinations, etc... Near all modern ATA, gateways, IP-Phones, Softphones are using delay before to send the complete numbered destination. The only problem could be a not so well built dialplan if you are using also legacy analog telephones behind Asterisk's FXS interfaces. What I was tempting to explain before (sorry if I was not so clear) is that it's possible to do as one wants, depending the target or volontee. This is why Asterisk is so powerfull and what I love it (realy) ;-) This was not to start a flamme war, sorry ! PS : hello to my friend Wilson ! Best Regards, Francois BERGERET, France. ___ --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] Upgrade 4 to 8 Analog Lines Question
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jim Freeze Envoyé : lundi 9 avril 2007 15:15 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Upgrade 4 to 8 Analog Lines Question Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase options as follows: 1) TDM40B - use with the current TDM40B 2) Sangoma Remora A20200 - use with the current TDM40B 3) Sangoma Remora A20400 - replace the current TDM40B Any info will be greatly appreciated. Thanks Jim -- Jim Freeze ___ --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 : RE : [asterisk-users] wireless desktop phones
Hi Tobias and the list, Yes, I have, I use and sell them to integrators ;-) But only the 600v3 family, not the older ISND or analog versions, and the current DECT handsets 40XX. Any Digium interfaces run well with them as any SIP IP-Phone, of course. The sound quality is GREAT and the infrastructure deployment possibilities wonderfull and scalable ! But, you must run a training with the company to well understand the how to do and capture the knowledge. I must also say that I am a radio guru and it's certainly easyer for me to understand this kind of equipments and how to avoid the deployment traps that an engineer who doesn't know what are radiocommunications but only VoIP. I have run them behind all current Asterisk versions, including the ASteriskNOW. Check about your codecs as usual. The last firmware from this last week suppresses few minor buggs occured during roaming in few previous cases. Best Regards, Francois BERGERET, France. -Message d'origine- De : Tobias Wolf [mailto:[EMAIL PROTECTED] Envoyé : jeudi 29 mars 2007 16:23 À : [EMAIL PROTECTED] Objet : Re: RE : [asterisk-users] wireless desktop phones [EMAIL PROTECTED] schrieb: Hi the list, Think Kirk solution ;-) www.kirktelecom.com Do you have this working in you enviroment ? Currently I have some test devices from Kirk (KIRK Wireless Server 600/3 with SIP protocoll and a couple of handsets). But i am not able to get audio between the handset and the destination then i call a zap channel. Calling another Kirk handset or another SIP phone (Snom) works quite well, then i dont put any options in the Dial Command. Otherwise i dont't get any audio also. Signalling a call is no problem. It would be great to hear from you if your setup work perfectly and what your enviroment is (Asterisk Version, type of Kirk Server). Thanks in advance, Tobias Wolf ___ --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] wireless desktop phones
Hi the list, Think Kirk solution ;-) www.kirktelecom.com This is an DECT/GAP infrastructure solution, and the bases can be seen as something like SIP/DECT gateways. Each wireless phone is like a separate IP phone from Asterisk side. You can use several bases and repeaters (only radio link, no Ethernet cable) to extend the range and have a global coverage into customers buildings. Very incredible, powerfull and scalable solution ! I think it's probably the only one with such a class and commercial grade. Best Regards, Francois BERGERET, France. ___ --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 and chan_misdn
Hi Pierre and the list, I have the habit to do like this after having compiled Zaptel and Libpri : cd /usr/src/ wget http://www.misdn.org/downloads/mISDN.tar.gz wget http://www.misdn.org/downloads/mISDNuser.tar.gz tar xzf mISDN.tar.gz tar xzf mISDNuser.tar.gz cd mISDN-1_1_1 make install cd ../mISDNuser-1_1_1 make install Move the modules which are in the bad directory : mkdir /lib/modules/`uname -r`/extra cp /lib/modules/extra/*.* /lib/modules/`uname -r`/extra Make the mISDN config files : /etc/init.d/misdn-init config Start mISDN : /etc/init.d/misdn-init start Go ahead and compile Asterisk : cd /usr/src/asterisk-1.4 ./configure make menuselect ; choose your options ! make;make install I hope this help ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Pierre Burton Envoyé : mardi 27 mars 2007 10:09 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk 1.4 and chan_misdn Hi, you also need mISDNuser. After that make clean make install you'll have access to chan_misdn. Regards. Pierre Administrator TOOTAI wrote: Hi list, I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went in zaptel dir and: make, make install, make b410p. Everything is ok. Now I want to compile Asterisk but can't activate the chan_misdn channel which depends on -from menuselect- isdnnet(E), misdn(E), suppserv(E) When I made the make b410p, all the misdn stuff was downloaded from digium's ftp. Also, running /etc/init.d/misdn-init --scan show me the 2 cards I have, /etc/init.d/misdn-init --config prepare me the misdn.conf and after a /etc/init.d/misdn-init start I see: mISDN_dsp 191656 0 mISDN_capi 88716 0 mISDN_l2 34452 0 mISDN_l1 11036 0 mISDN_core 71360 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 kernelcapi 44576 2 mISDN_capi,capi My questions: why Asterisk doesn't want to let me activate the misdn channel? Is misdn ready for 1.4? Thanks for any hint ___ --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] Two or More Bri Cards
Hi ! Prefer to have only one card with how many ports you want. Always better for IRQ flow. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed Envoyé : lundi 26 mars 2007 09:11 À : asterisk-bsd@lists.digium.com Cc : asterisk-users@lists.digium.com Objet : [asterisk-users] Two or More Bri Cards hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq -- ___ --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] SIP/IAX peers UNREACHABLE and audio loss
Hi men, I have already encountered some issue like this with few switches (very known great brand) which doesn't like VoIP traffic ! Check by drectly connected the VoIP equipment - if you can - with temporary long Ethernet cables bypassing the tested switch to see what happens in this case. You can also tell to qualify with a longer delay, but this could not help in case of regulary frames losses. Good luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rajeev Natarajan Envoyé : samedi 24 mars 2007 08:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. Rajeev On 3/23/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --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 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] Issue with Hamlet ISDN PCI card(Cologne Chipset)
Hi David and the list, It's normal ;-) Near all European BRI operators cut off the line between calls. So, you must trieve the correct parameter avoiding to survey the line as for mISDN : pmp_l1_check=no I use mISDN without any issue with B410P. I hope this help. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Francesco Peeters (Asterisk) Envoyé : samedi 24 mars 2007 12:40 À : Asterisk Users Mailing List - Non-Commercial Discussion Cc : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset) On Sat, March 24, 2007 11:54, Mauro Zanin wrote: Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Not really... It's a bristuff quirk... It doesn't gracefully handle the forced D-channel down that most European ISDN operators implement. That is why I switched to testing vISDN, but that has been stagnant for over half a year without any fixes for a few very annoying bugs, because the programmer dedicated all his time to rewriting the vGSM part... I am now testing mISDN as someone on the vISDN list mentioned that it's chan_misdn voice support had greatly improved... The only way I can *somewhat* keep bristuff working without contacting the ISDN carrier to turn on the D channel permanently is by initiation a 100ms outbound call every minute using the manager interface... (Yes, a very ugly kludge indeed, but I do not want permanent channel up, as I want to be able to test everything in a normal environment, as I am planning to install this in other location too once I have a stable, reliable environment) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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: RE : SIP/IAX peers UNREACHABLE and audio loss
Have you taken care of any eventual IRQ sharing ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Edoardo Serra Envoyé : samedi 24 mars 2007 20:27 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss Martin Joseph ha scritto: The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. It was also my first though, but my LAN is very SIMPLE, so I was wondering if something else could cause the problem. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. You're probably right, but it depends on Asterisk internals (which I don't know well). If Asterisk would stop to send RTP audio when just a qualify packet get lost it can make the situation worst. If the asterisk servers are all on your LAN then the network issue should be easily fixable. It should, but my LAN is very simple... I have a 10/100 Mbit switch with no more than 15 servers on it. Traffic on the LAN is not heavy even if the time of the day I see in the logs make me think it could be an issue related to network load trafic Anyhow I'll try to generate some heavy traffic on the LAN to see if it could be related to that. I also noticed that this problem began to happen when I upgraded my Asterisk to 1.2, but it can be a concidence. Do you think it could be related to bugs in ethernet drivers, kernel or whatever at the OS level ?? If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. We have some Asterisk spread all over the public Internet, but firstly we should solve this problem at a LAN level Tnx for attention Regards Marty ___ --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 ___ --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] TDM2400 Hardware Echo Cancel
Check without the echocan module (remove it) if any 'crackle is listen again. If yes, the echocan is not faulty. If yes, check another echocan module temporary. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ed W Envoyé : jeudi 18 janvier 2007 12:39 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TDM2400 Hardware Echo Cancel Hi Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. Just a thought, but check that your gain levels are not too high? Ed ___ --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] 5v capable motherboards
Hi Mark and the list, You can switch to Industrial PC. I mainly use PICMG 1.0 standard Single Board Computer cards and passive PCI busses with success. They are 5V PCI bus and maintained for ten's years as industrials want. Perfect for IPBX with a long live or MTBF. I prefer Pentium-M equiped SBC for low consumption and low profile and very low noise CPU fan. This permits to save extra power for cards which sucks many watts during FXSX ringing. Their components are high class level regarding many motherboard for low cost production. Good luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mark Farver Envoyé : vendredi 12 janvier 2007 19:38 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] 5v capable motherboards Anyone have a suggestion on where I can get a decent new MB with 5v capable PCI slots. It seems like every decent server MB on the market has 3.3V slots only. Is diving into the junkbin my only choice if I can't afford to replace the 5v quad-T1 wildcard? Thanks Mark Farver ___ --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] TDM2400p bad sound quality
Hello ! Shared IRQ ? Very old tired CPU ? No echocan module on the TDM2400 (echocan Zaptel solution claims more motherboard CPU power) ? Not enougth RAM ? Not CPU optimized compilation with 1.2 ? Please describe more your server and Asterisk version... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Giuffredi Envoyé : vendredi 12 janvier 2007 19:16 À : asterisk-users@lists.digium.com Objet : [asterisk-users] TDM2400p bad sound quality Hi list, I have this problem: when someone is making a call, with asterisk and a TDM2400P connected to 8 fxo lines, the sound is good, but if three, for people are calling at the same time the sound got worse and worse. Using other voip cards the sound is much better even with all user calling at the same time. What can be, the problem? Someone else has having the same issue? 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
RE : [asterisk-users] Happy 2007!!!
I wish many stars in your blue sky for this new year :-) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Sam Tam Envoyé : dimanche 31 décembre 2006 19:19 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [asterisk-users] Happy 2007!!! Happy New Year .. Sam _ From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Monday, January 01, 2007 12:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Happy 2007!!! Its the new year. Cant we all be semi nice for atleast a lil bit ? - Original Message - From: Jason Parker mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Sunday, December 31, 2006 10:48 AM Subject: Re: [asterisk-users] Happy 2007!!! I haven't quite figured out what he's selling though.. - Original Message - From: Tom Lynn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 30, 2006 7:59:12 PM GMT-0600 US/Central Subject: Re: [asterisk-users] Happy 2007!!! Sounds like an EBay ad... On 12/30/06, Josué Conti [EMAIL PROTECTED] wrote: Always... Desire that in the New Year that if you really initiate... It hears the words that always it desired to hear. It pronounces the phrases that one day it desired to repeat. It feels the emotion that always waited to feel. It walks for the tracks that one day it desired to follow. It divides the affection with who always desired to distribute. It hugs all the friends whom always it desired to congregate, and alive the life that always dreamed to exist... Happy 2007 Best Regards Josué ___ --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 -- Jason Parker Digium _ ___ --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] TE110P with Qsig
Hi Josué, Have you checked the strap on the TE110P board ? You must have it on the E1 position, not T1 (open ?, I don't remember at this hour, sorry). Check also without crc4. And recheck ztcfg -vvv. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Josué Conti Envoyé : vendredi 29 décembre 2006 23:27 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TE110P with Qsig Hi Matthew thank's will be attention. I believe that the configurations are correct, I changed of server, one another hardware and the problem remains the same. :( Changing of protocol, for euroisdn the problem remains. Stranger, does not find? Best Regards Josue zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=us defaultzone=us zapata.conf [trunkgroups] [channels] language=us context=default switchtype=qsig nsf=none pridialplan=unknown prilocaldialplan=unknown facilityenable = yes signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes restrictcid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 immediate=no callerid=asreceived musiconhold=default group=1 channel=1-15 channel=17-31 2006/12/29, Matthew Fredrickson [EMAIL PROTECTED]: It sounds like it isn't configured correctly. Are you sure that your cabling is ok and that your span= line is correct? Matthew Fredrickson On Dec 28, 2006, at 8:29 PM, Josué Conti wrote: Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. === == Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of
RE : [Asterisk-Users] asterisk + door opener
Hello the list, You can use FXS and em signalling to reverse the line polarity temporary to trigger an external door opener interface. This is very easy. Good Luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Thomas Kenyon Envoyé : jeudi 21 décembre 2006 12:13e. À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] asterisk + door opener Jerry wrote: Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. I'd be interested too, I was thinking of upgrading our door opener with a telephone line adapter and an FXO port from the linecard, but if I can do this without using an FXO port (and doesn't cost the earth) It would be great. ___ --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] Linux distro + Asterisk or Trixbox?
Hi men, Have a look at : www.asterisknow.org This will be THE standard ! Best Regards, Francois BERGERET, France. ___ --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: RE : Re: Recommendation for FXO
Hi Marty, I have checked/played FXS ports behind Asterisk with success and checking now a new firmware for FXO one stage (normaly two stages). All this gateways have the same manager unit and parameters suite, looking like Cisco models. It's normaly easy to use if you have trained for Cisco (only few differences). This is a real manufacturer, not a copy provider, and all the 19 units have the same look for near all the brands. I know FXS models with 32 ports running well for near to one year now. I have few pictures if you want to see the PCB's... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 4 décembre 2006 21:58 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: RE : Re: Recommendation for FXO On 2006-12-04 11:54:14 -0800, [EMAIL PROTECTED] said: Hi the list and Marty, Take a look to www.aliwei.com. Thanks for the idea, but this looks CHILLINGLY identical to the wellgate? I wonder if this is the same hardware with a different name on it? Is this something you are using personally (in particular the FXO and and asterisk)? Thanks again, Marty Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 4 décembre 2006 20:47 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: Recommendation for FXO On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said: snip So, I would like to purchase another PSTN gateway which WORKS WELL with asterisk. I need it to hook up via ethernet, since my platform of choice (mac OSX) has no PCI card support. I only have one PSTN line, and already have other ATA's for FXSs, so I really only need one FXO port, although I realize there is no such animal. Any positive experiences with FXO gateways that connect via ethernet? Especially with a long loop/echo issues (ie not SPA3000)? I am wondering if anyone has experience with the Audiocodes MP114 (2fxs/2fxo)? This is pricey, but I am SO sick of mucking around with consumer grade bs (ie grandstream). I am also curious about the mediatrix 1204 and the Multitech MVP-130 although it sort of looks to me like the multitech doesn't do SIP. Any thoughts or help, before I take an expensive leap? Marty PS asterisk 1.2.13 ___ --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 ___ --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: Recommendation for FXO
Hi the list and Marty, Take a look to www.aliwei.com. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 4 décembre 2006 20:47 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: Recommendation for FXO On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said: snip So, I would like to purchase another PSTN gateway which WORKS WELL with asterisk. I need it to hook up via ethernet, since my platform of choice (mac OSX) has no PCI card support. I only have one PSTN line, and already have other ATA's for FXSs, so I really only need one FXO port, although I realize there is no such animal. Any positive experiences with FXO gateways that connect via ethernet? Especially with a long loop/echo issues (ie not SPA3000)? I am wondering if anyone has experience with the Audiocodes MP114 (2fxs/2fxo)? This is pricey, but I am SO sick of mucking around with consumer grade bs (ie grandstream). I am also curious about the mediatrix 1204 and the Multitech MVP-130 although it sort of looks to me like the multitech doesn't do SIP. Any thoughts or help, before I take an expensive leap? Marty PS asterisk 1.2.13 ___ --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] mISDN
Hi the list,
You must input extensions using the 5 (may be 4 in some countries) last
digits representing your telephone number end for this BRI line in your
current ISDN calls incoming context.
Open the ISND debug mode and see what is on your asterisk console screen
when a call comes.
That's all ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Timothy Parez
Envoyé : mercredi 29 novembre 2006 13:27
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] mISDN
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten = _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten = _.,2,LookupCIDName
;exten = _NXXNXX,3,Dial(sip/sammy,30,r)
;exten = h,1,HangUp()
;exten = s,1,Dial(SIP/timothy)
;exten = s,2,Hangup()
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN}) exten
s,2,Answer() exten s,3,Echo() exten s,4,Hangup() exten i,1,NoOp(Invalid call
from ${CALLERID} for ${EXTEN}) exten i,2,Answer() exten i,3,Echo() exten
i,4,Hangup()
As you can see I tried a few things, but none of them work.
Does anybody know how to solve this ?
Thnx.
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RE : [asterisk-users] Fxo box for asterisk ?
Hello, All the biggest gateways manufacturers do that. Search for Aliwei, Audiocodes, Patton, etc... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Noc Phibee Envoyé : lundi 30 octobre 2006 20:51 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Fxo box for asterisk ? Hi do you know if they have external Box (not internal card) for connect Analog Line and Pri/Isdn to asterisk for incomming and outgoing calls ... 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 ___ --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] New Asterisk StumbleUpon Group
Hello Matt, I have not seen how to add a site. Could you help me (us) ? Tks Francois Bergeret, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Matt Riddell (IT) Envoyé : vendredi 6 octobre 2006 11:40 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] New Asterisk StumbleUpon Group -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Just thought I'd let people know that I've created a new StumbleUpon group for Asterisk sites. If you have a site that is related to Asterisk and is not listed, feel free to add it. Alternatively, if you're new to Asterisk and want to find out what sites are out there pop on over and have a look: http://asterisk.group.stumbleupon.com/ - -- Cheers, Matt Riddell ___ ___ --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] TDM2400P wiring.
Hello, RING 1 26 TIPfirst Zap channel RING 2 27 TIPsecond Zap channel RING 3 28 TIPthird Zap channel etc.. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de C F Envoyé : mardi 3 octobre 2006 06:00 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] TDM2400P wiring. I just received my first TDM2400 card I tried searching and couldn't find anything on this. I have 2 FXO modules with this card, it came with one modlule in the slot marked as slot 6, so I put the other in slot 5. Since I don't have an Amphenol connector/cable and a 66 block at the moment I can't realy test it. I'm therefore turning here for help. Which slot on the TDM24xxP card is Pair 1 thru 4 on the 66 block? 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] University switches to Asterisk
Eric, contact me off list and I will give you a nce exemple with a worldwide Asterisk network ;-) Francois BERGERET, France. f6hqz-m_at_hamwlan.net -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : jeudi 14 septembre 2006 17:21 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] University switches to Asterisk That is not helpful in convincing my customers that there are many companies using Asterisk. Michael Welter wrote: Yes. I don't use my customer's names on the list, so I can't say anything. Porier, Jeremy M. wrote: They're not the only ones :-) Jeremy Porier Senior Director of Information Systems and Technology Colorado Christian University [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 13, 2006 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] University switches to Asterisk Interesting article I found linked from Groklaw: Sam Houston State University replaces Cisco CallManagers, Nortel PBXs with Linux-based VoIP and messaging servers http://www.networkworld.com/news/2006/091206-von-sam-houston.html?pag e=1 Doug ___ --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] Problem with a TDM400P
Hello Mark and the list, What about if you change the order of the modules, starting with FXS first and finishing with FXO on the TDM400P slots ? I remember to have read something like always start with FXS if FXS and FXO modules are present on the board... Feedback please. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mark Muffett Envoyé : lundi 28 août 2006 19:49 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Problem with a TDM400P I'm setting up my first (and very simple) Asterisk PBX and running into problems with the FXO module I have on a TDM400P - I'm trying to connect to a standard UK, BT, POT. The problem is that when I plug the FXO module into a functioning BT line, it seems to make the line become engaged - ie if I try to call it from another number I just get the engaged tone. This happens whether or not asterisk is running and even whether or not the zaptel modules are loaded. The TDM400P card seems to be ok - I get the expected line: 04:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when I type lspci. My FXO module is in position 1, with FXS modules in positions 2 and 3. The FXS modules seem to work ok with my config files (I get a dialing tone if I connect a phone to them). My zaptel.conf file is simply: fxsks=1 fxoks=2,3 loadzone=uk defaultzone=uk and my zapata.conf is (at the moment): [channels] ; context=test usecallerid=yes hidecallerid=no immediate=no signalling=fxo_ks echocancel=yes group=1 channel=2 channel=3 signalling=fxs_ks echocancel=yes busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes group=2 channel=1 (I've tried getting rid of busydetect, answer/hanguponpolarityswitch, and callprogress individually and all together). Could I have a hardware fault? - if so any ideas what tests to run? Or is there something else I need to configure. Thanks for any help. Mark ___ --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] Connecting branch offices through IPsec tunnel --latency effects?
Hi Stephen, +99 ms via IPSec FreeSWan But good protection and no NAT issue. Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Stephen Bosch Envoyé : mardi 25 juillet 2006 17:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects? Hi: If I connect two offices through an IPsec tunnel, what is the impact on latency, and does it noticeably affect calls? Has anyone out there tried this? What were the effects? Cheers, -Stephen- ___ --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] X100P clone not working
Hi Franck, NOACPI and the sound must be more clear. And, of course, have you tell to /usr/src/zaptel/zconfig.h and /usr/src/asterisk/Makefil what kind of processor you have and enabled MMX if possible before to compile ? Good Luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Frank Darner Envoyé : dimanche 23 juillet 2006 23:41 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] X100P clone not working Am Sunday 23 July 2006 20:58 schrieb Walter Willis: look udev rules??? the problem was related that ztcfg did not find zaptel.com -c /etc/asterisk/zaptel.conf has solved this issue #ztcfg --help -c filename -- Use filename instead of /etc/zaptel.conf my failure, I should read man page more carefully now am trying to get it working, the sound is still unreliable ___ --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] Running 40 active calls (too much för CPU?)
Hi ! Call Digium crew. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : mardi 4 juillet 2006 11:20 À : asterisk-users@lists.digium.com Objet : SV: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hello again, I read this interesting article about the TE405P card. How do I check what firmware version my card has? http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html ... And how do I update it if it's an old one? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --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 ___ --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] x100p buying advice
Hello, Ridiculous business argumentation... By changing 2 resistors maping on the same card you can say to system that is any response as X100P, X101P, or Clone. No proof to good quality or if it realy run ! Take a look to voip-info.org about X100P and X101P, you will learn more about the chipsets, which is only the available information. Use this kind of cards (WinModem) for tests or curses only, no production. Good Luck. Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rod Morison Envoyé : mardi 27 juin 2006 02:16 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] x100p buying advice I'm looking to get an x100p off ebay and am not particularly familar with the life cycle of the card.. An Authentic X100P listing has a buy it now of $29.95 and says There are 3 types of cards Asterisk would recognize: *Screenshots from the official, original driver install Cheap OEM X100P,Clones, Compatibles, Knock-Offs Found a Wildcard FXO: Generic Clone The X101P (note the 101, not 100) is a Low-end version of X100P which uses low grade chips Found a Wildcard FXO: X101P Authentic, Original X100P Speaks for Itself! Found a Wildcard FXO: X100P From what I gather clone's and knockoffs will have trouble with callerid. Is the Found a Wildcard FXO: X100P enough to establish full featured hardware (assuming an honest seller)? Is there another recommended source besides ebay in this price range? 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 ___ --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] quad t1 / 1U rack server combos
Hy men, :-) Use Industrial PICMG PC's. Higher cost at buy, but very stable and evolutive platforms. SBC doesn't change during a long industrial period. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Totaro Envoyé : lundi 12 juin 2006 01:06 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] quad t1 / 1U rack server combos Colin Anderson wrote: C'mon guys! Certify a few current model servers and be done with it. Problem is, certification is a moving target and can become invalid with something as simple as a BIOS change by the manufacturer. Now that the barrier to entry to changing a design is almost nil, manufacturers love to screw around with designs to save a few bucks. I have seen two identical boxes, labelled as such by the manufacturer, bought in the same time period, but with different guts. Digium would wind up with egg on their faces by certifying a system, then 90 days later after everyone buys it, finds out that some subtle change by the manufacturer has destabilized the config. I agree it is frustrating as hell, but this is the price we pay. Would you rather buy a Mitel for 10X the $$$? Maybe in some circumstances, it is worth it. -- -- lman/listinfo/asterisk-users I bought two HP DL380s at the exact same time from CDW. I used the first one to build an image to transfer to the other system. When I booted the second system, kudzu reported different NICs. So, yes, next to impossible to certify any hardware without the hardware manufacturer's help. ___ --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] Need a recomendations and config samples.FXS-SIP terminal with 4 ports.
Hello, I use and sale (as Distributor) Micronet and Aliwei gateways. Fine and stable, without echo. Each port is seen as a separate SIP account. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Nikolay Pavlov Envoyé : vendredi 26 mai 2006 15:15 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Need a recomendations and config samples.FXS-SIP terminal with 4 ports. Hi, folks. I want to buy FXS-SIP terminal with 4 ports (up to 250$). Do you have any recomendations and Asterisk configurations samples for such devices. Any pitfalls? Actually i realy don't know what to buy? -- = = Best regards, Nikolay Pavlov. = = ___ --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] TDM2400P with echo canceller not working
Hello Giorgio, I am a TDM2400 happy user. :-) Could you show your zaptel.conf zapata.conf config files ? Think to tell us how many modules you have and where they are plugged on the TDM2400P. Are the leds on the echocan modules running as a LasVegas casino (scrolling in a circular pattern) ? If you have an echocan module aboard and well running you must see it at Linux boot (syslog): The 4 echocan chipsets are sequencialy checked and return an hexadecimal code different of FF if ok, just before to tell VPM: Present and operational (Rev X), if I remember well. If the VPM is ok, zaptel echocan software is automaticaly disabled for this zap channels in Asterisk server. The echotraining is not applicable for this card and generate an error message on the Asterisk console if enabled in your zapata.conf, but don't care, this doesn't affect Asterisk and the call itself. The settings must be done as usual with TDM400 cards. I hope this can help a little. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Giorgio Incantalupo Envoyé : lundi 29 mai 2006 09:33 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] TDM2400P with echo canceller not working Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found no info about how to use this card and how to correctly set zapata.conf, which zaptel version to use, etc... Does anybody knows how to use this card? TIA Giorgio Incantalupo ___ --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] PCI Problems
Hi the list ! I share Ethernet card IRQ with my TDM2400 without any trouble here, on an old Intel motherboard and an old PII400 ! This is another proof that sharing IRQ is not necessary an issue. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Andrew Kohlsmith Envoyé : vendredi 26 mai 2006 16:37 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] PCI Problems On Thursday 25 May 2006 16:11, Sean Cook wrote: What could be the other causes? I have exhausted everything I know how to do. PCI sharing explains it (whether or not it is infact the problem). This card shares the BIOS assigned interrupt with the network card... Audio problems can come for a variety of reasons. They are caused by (but not limited to) things such as - IRQ sharing with another device with a shitty driver or poor hardware - Poor/inconsistent PCI bus behaviour and timing - overloaded CPU or poor kernel parameters which cause timing problems - shitty hardware or drivers which can lock out IRQs for a long time - buggy drivers for the TDM or ethernet hardware - bad PCI tuning with setpci or kernel parameters, latency timers especially - other hardware (PCI bus controller, north or south bridge) issues - faulty hardware - poor cabling (either TDM side or ethernet side) IRQ sharing is often blamed for audio problems but the fact of the matter is that IRQ sharing is *NOT* an issue if the hardware that is sharing the IRQ (and the drivers for that hardware) plays nicely and reacts to the IRQ quickly. PCI is DESIGNED to share IRQs. The trouble comes when vendors take old ISA hardware, port it to PCI and/or don't ensure that they not only share IRQs properly but also do not ensure that their drivers check that their hardware caused the IRQ and react to IRQs quickly. There is NOTHING inherently wrong with sharing IRQs. The IRQ handler needs to check the hardware to see if it was their hardware that generated the IRQ and get the hell out if not. A lot of (poor) drivers do NOT do this. The driver either assumes that the IRQ MUST have been generated by the hardware (which can cause a host of weird problems), or the check takes so long that it causes trouble for the card that DID generate the IRQ. Digium's hardware is more sensitive to IRQ sharing trouble than other hardware for two very simple reasons. The first is that the TDM cards have no real buffering. If the data is not taken from the register it will quickly be overwritten by the next block of data. This is analogous to the old 16450 UARTs of yore. They had a receiver shift register and a 1-byte receiver buffer. If you didn't get the data out of the buffer before the next byte had shifted in, the new byte would be transferred to the buffer and you'd get an overrun error. The 16550 replaced the 1-byte receive buffer with a 16-byte FIFO (IIRC) -- you could trigger an IRQ after the FIFO had filled 'x' bytes, and then service the IRQ, retrieving all bytes received in one fell swoop. And if your IRQ service routine got a little delayed it was no big deal because there was room for another byte or two before you started losing data. This allowed the IRQ volume on busy serial applications to be far lower (up to 16x lower) than before, which allowed for better system utilization. Digium's hardware is like the old 16450. There is no FIFO. This was done consciously, and is not necessarily a bad design -- TDM is VERY sensitive to latencies. The more delay you have, the worse things like echo become. Bringing TDM data into the PC is already pretty laggy. Adding more delay with FIFOs isn't necessarily a good thing. (I would argue that having a 16 byte FIFO and triggering the IRQ on the first position would not be a bad thing nor would it introduce any latency, but that's me. I'd change a few things about Digium's hardware, but there is no arguing at their success.) So back to the problem at hand: if there is significant delay between the IRQ and the IRQ service, you lose data. This leads to chirping/clicking and in the case of T1, HDLC/framing errors, dropped links and bouncing D channels (for PRI). The second reason is that Digium's drivers do a LOT of work in the IRQ handler. Essentially they are poor PCI neighbours. In the past (I have not checked this recently) all of the echo cancellation and heavy lifting was done right inside the IRQ handler, with interrupts disabled. This caused their IRQ service time to be lengthy, and until interrupts are enabled again you essentially lock out any other driver from servicing its hardware. (Basically Digium's drivers do to other drivers what Digium's drivers can't stand to have done to it.) Contrast this with Sangoma's drivers, which get the data into system RAM, set a flag (softIRQ?) and then get the hell out of the IRQ context as quickly as possible. Then whenever the CPU
RE : [Asterisk-Users] Problem with a TDM-400P
Hello, Check your gren module by moving it from slot to slot on the TDM400P card. If the problem is following your module, it's the module itself the cause. If not, and running well on other slot, it's the TDM400P itself. Good Luck ! Best Regards, Francois BERGERET, France. ___ --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] dialing FXO gives wrong billsec
Hello Yusuf, This is a normal use of zap channels : it is not possible to see if the call is realy answered, and Asterisk say yes as soon as the call is placed. That's all... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Yusuf Envoyé : mercredi 3 mai 2006 23:08 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] dialing FXO gives wrong billsec Hi all, I came across a new(to me that is) issue. I want to know from others what they have done to resolve this. I have a 4 port digium card with FXO's, and connected to each FXO is a premicell. When I dial the premicell, after about two seconds is says 'ZAP/1 answered', then it takes a few more seconds for the call to hit the cellular network, before the cellphone starts to ring. However, asterisk sees the 'ZAP/1 answered' as the cell phone being answered, so my billsecs in the cdr's are off by ten seconds or so, and all the cdr's are 'ANSWERED', even though the cell phone was not answered. my dial string looks like so: (all calls come in to inbound) [inbound] exten = _X.,1,Dial(ZAP/1) I have a standard zaptel and zapata, Asterisk 1.2.6 thanks, yusuf -- 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 ___ --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] IAX Configuration
exten = 19,1,Dial(SIP/19,20,tr) Must be : exten = 19,1,Dial(IAX2/19,20,tr) Because you are using IAX IP-Phones... Best Regards, Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Olivier Saulnier Envoyé : mardi 2 mai 2006 16:09 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] IAX Configuration Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten = 19,1,Dial(SIP/19,20,tr) exten = 19,2,Voicemail(u19) exten = 19,hangup exten = 19,102, Voicemail (b19) exten = 19,103,Hangup exten = 20,1,Dial(SIP/20,20,tr) exten = 20,2,Voicemail(u20) exten = 20,hangup exten = 20,102, Voicemail (b20) exten = 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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 ___ --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] TE410P card connection (was: Pinouts for T1/E1crossover cable)
Hi Louis-David, Check without crc4 Best Regards, Francois BERGERET. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Louis-David Mitterrand Envoyé : dimanche 23 avril 2006 09:39 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] TE410P card connection (was: Pinouts for T1/E1crossover cable) On Sat, Apr 22, 2006 at 11:59:21AM -0400, Alexander Lopez wrote: Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet. Thanks for the info! Should I use a T1 cross cable to connect the telco's socket to the TE410P card? When I tried straight cat5 cables, both leds remained red at each end. However this E1 socket works fine with the Matra PBX, so it must be a cable problem or TE410P misconfiguration. Thanks, Here is my configuration: /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone=fr defaultzone=fr /etc/asterisk/zapata.conf: ;; to telco context=default signalling=pri_cpe group = 1 channel = 1-15 channel = 17-31 ;; to old pbx context=international signalling=pri_net group = 2 channel = 32-46 channel = 48-62 ___ --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] How do I limit the lenght of a call
Hi John, If you enter show application dial when logged into the Asterisk console, you can read that help (extract only regarding dial option) : L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option: * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE yes|no Play sounds to the callee. * LIMIT_TIMEOUT_FILE File to play when time is up. * LIMIT_CONNECT_FILE File to play when call begins. * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. The default is to say the time remaining. Always think to display help through the show application ..., it could be pertinent. Good luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de John Rich Envoyé : dimanche 16 avril 2006 15:46 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] How do I limit the lenght of a call Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] Echo cancellation
Hi, zap show channel 5 To see channel 5 specs, and take a look at Echo Cancellation: 128 taps unless TDM bridged, currently OFF during calls, you must have ON. If you have hardware echocan module, as for TDM2400E, you must also read DSP: yes if this module is active. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Giordano Grandis Envoyé : mardi 28 mars 2006 16:54 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : R: [Asterisk-Users] Echo cancellation Ok, but is there a way to check if echo cancellation is active on a call in progress ? Thanks Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Steve Davies Inviato: martedì 28 marzo 2006 16.43 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Echo cancellation On 3/28/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, I'm using bristuff 0.2.0 RC8o with a HFC pci card and on several calls I saw that the echo cancellation is on OFF Echo Cancellation: 0 taps, currently OFF (the result of zap show channel 1-1 for example) Echo cancelling is only enabled if there is a call in progress. Cheers, 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 ___ --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 : RE : [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)
This card doesn't permit to support Mark Spencer's company and project. This card has no hardware echocan and use only the X100M and S110M clones modules. This two reason are sufficient for me. -Message d'origine- De : Krzysztof Drewicz [mailto:[EMAIL PROTECTED] Envoyé : lundi 27 mars 2006 21:45 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines) [EMAIL PROTECTED] wrote: Hi, Jump to a TDM2402E for 6 POTS lines with hardware echocan. Only one IRQ used, and easy future extensions by adding modules. Have anyone here used a clone i.e. A1200P-01 (A1200P + 1 FXO100 module) ? ___ --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] FXS channel banks
smime.p7m Description: S/MIME encrypted message ___ --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 : RE : [Asterisk-Users] FXS channel banks
2 x TDM2460E (with hardware echocan module) or TDM2460B (wo/echocan) = 48 phones lines, no T1 cards, no channel banks level adjustments troubles, direct Zap channels and simple switching. Probably the best choice and price :-) Best Regards, Francois BERGERET, France. A very happy TDM2400 user ;-) ___ --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] Stability of Asterisk with 2 x TDM400P cards (6analogue lines)
Hi, Jump to a TDM2402E for 6 POTS lines with hardware echocan. Only one IRQ used, and easy future extensions by adding modules. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jared Davison Envoyé : vendredi 24 mars 2006 05:26 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6analogue lines) I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY stability of the Asterisk system associated interface hardware and drivers. Do people need to reboot their systems regularly? Thanks in advance. Jared ___ --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] FXS channel banks
Title: Message How many phones lines ? -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Curt ShafferEnvoyé: vendredi 24 mars 2006 03:17À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] FXS channel banks Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt ___ --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 : RE : [asterisk-dev] iax failure?
Oops ! I have upgraded TRUNK again via SVN and all was seeming to be fine, no more invalid IAX2 frames and able to place and receive calls. I was happy.. But, few calls later (about 5 minutes) : INVAL frames again and no more possibility to place or receive calls, no prompt tone, nothing ! Strange... Best Regards, Francois BERGERET, France. ___ --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] Sipura 3000 DMTF
Check for : dtmfmode=outband Good luck ! Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Chris Mason (Lists) Envoyé : samedi 18 mars 2006 17:43 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Sipura 3000 DMTF I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband disallow=all allow=ulaw [3200] type=friend host=dynamic context=pstn-in secret=mysecret qualify=yes dtmfmode=inband disallow=all allow=ulaw insecure=very [pstn-spa3k1] type=peer auth=md5 host=192.168.101.11 port=5061 secret=mysecret username=asterisk fromuser=asterisk dtmfmode=inband context=pstn-in insecure=very -- Chris Mason NetConcepts (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 ___ --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] TDM 2400 With 24 FXO
Hello Fernando, I have checked this card with and without hardware echocan : the hardware echocan module does the job better than the zaptel software can do it. I recommand this module without any doubt. But, the echocan algorithms in zaptel are better and better and the CPUs power grows permanently. It is possible to use this card without hardware echocan, but you will encounter the same results, in this case, as you can obtain with the other TDM Digium's cards : correct for certain situations, not for all extreme cases, depending what listening level your users want, lines specifications and what critical echo threshold they can admit before to not be able to do correctly their job. Near same thing for E1/T1 harware echocan features. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Fernando BERRETTA Envoyé : vendredi 17 mars 2006 14:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] TDM 2400 With 24 FXO Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando ___ --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] Asterisk Users Mailing List Traffic
Of course, but if newbies are separated and together only without any expert, who can explain them anything ? I am actualy a subscriber for all the Digium lists. If more lists will be, more subscribtions I will get and I will receive the same quantity of messages ;-) Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : samedi 18 mars 2006 23:05 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk Users Mailing List Traffic I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --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 ___ --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] Best budget IP phone at the moment?
Hi, Check Chinese IP-Phones with PA1688 chipset : IAX2/SIP/H323/MGCP/Net2Phone + all the main codecs ! Good Luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de WipeOut Envoyé : vendredi 17 mars 2006 13:11 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [Asterisk-Users] Best budget IP phone at the moment? Hi, I am looking for a budget IP phone that can use preferably iLBC or GSM codecs.. Suggestions? ___ --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] IAX Phone?
Hello, As I have said earlier in the list, take a look at Chinese IP-Phones with PA1688 chipset : IAX2/SIP/H323/MGCP/Net2Phone + all the main codecs ! Good Luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Joe Hood Envoyé : vendredi 17 mars 2006 20:37 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] IAX Phone? Is there such a thing? Or is the Digium IAXy device the closest one can come? Additionally, any idea how to get the message waiting light to work through Digium IAXy device? Thanks, Joe ___ --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] Echo Cancellation
Hi Asterisk's people, You can buy Digium's card harware echo can models without the echo can module and buy it later if necessary. They are scalable ;-) Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de mustardman29 Envoyé : mardi 14 mars 2006 22:24 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] Echo Cancellation The way I look at it, it's better to have hardware echo can and not need it than to need it and not have it. The cards are not upgradeable to hardware echo can and it is almost impossible to pre-determine if you will need it. Software echo can sometimes does a good enough job but it will never be as good as hardware. You get what you pay for. -Original Message- From: Keith Schmidt [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 14, 2006 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Echo Cancellation I have 3 POTS lines that I want to use with Asterisk, I am looking at prices for FXO cards and the cards with echo cancellation are really pricey... is echo cancellation really worth it for a 3 or 4 line system? Will I notice a difference without the echo cancellation? Thanks Keith Schmidt ___ --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] Voice problem
Title: Message Hello, Have you optimized by chosing the correct CPU and see for MMX support before to compile Zaptel and Asterisk ? What is your server cofiguration ? How is its load ? How many simultaneous calls ? Etc... All litle details which can help to consider and understand your problem is welcome ;-) Best Regards, Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Andrew NowrotEnvoyé: dimanche 12 mars 2006 12:59À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] Voice problemHi,I have small issue with Asterisk. My customers complaining that sometimes (not always) the outgoing voice (the voice which can be heard by the user a the other end) quality is very low (stutter and sudden clicks). The problem exist in only-IP configuration and in IPtoTDM connections as well. I use alaw codecs. I know that they consume a lot of bandwidth, but the upload and download stream is about 1Mbit/s so the voice problem can't be cause by the lack of bandwidth. Does anyone meet something like that? CheersAndrew ___ --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] T1 GSM or DECT ? Searching for a complete microcell solution
Hi gentlemen :-) I am searching a radio base GSM or DECT with high power for long range, and the terminal units (handy). This equipment must be connected to a T1 port from an Asterisk. The number of simultaneous channels must be 7 to 10. Do you know a manufacturer with nice equipments at correct price ? Thanks in Advance. Best Regards, Francois BERGERET, France. ___ --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] TDM11B Hang up detection not working in France ?
Hi Pascal ! France is not more difficult than other country. This is one of my channels behind France Telecom : usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=11 ; definitive level for no loss -2 dB txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no adsi=yes busydetect=yes busycount=3 busypattern=500,500 signalling = fxs_ks callerid = asreceived amaflags = documentation context=WHAT_YOU_WANT channel = 6; my current channel number for this setting I hope this could help you and some other french guys. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Pascal OFFREDO Envoyé : jeudi 9 mars 2006 16:59 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] TDM11B Hang up detection not working in France ? Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount, hanguponpolarityswitch) without success. I've googled around and it seems this problem is specific to France. Is there any French people in this list that has a TDM11B that hangs up correctly ? regards Pascal OFFREDO ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.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 ___ --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] Ringing Delay
Hi Chan, 1/ be sure to have correctly inputed your country zone 2/ disable the fax recognition in zapata.conf Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de chan (Alpha Trilogies Networls) Envoyé : lundi 27 février 2006 08:35 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Ringing Delay Hi, Can some one advice me that how can I make the FXO channels port answer an incoming calls, means when I call from Lan line to Asterisk TDM400, my phone get ring immediately. When POT FXO port is ringing, Asterisk seems like studying the incoming ringing pattern even it did answer the call. I did not activate the usedestingtive, but why it seems delaying an incoming calls? Normal PBX, say will only delay 1 cycle as max in analog line, but Asterisk is about 2 sec...??? ___ --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] IAX2 through Shorewall rpoblem
Hello the list, Be carefull to have this rule available at begining of your rules list, because shorewall use the first one matching and stop to check the following. If you have another with a range including this UDP 4569 DNAT before your new one (as UDP 1024 to 65535 for example), it could shortcut it definitively... Best Regards, Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rich Adamson Envoyé : jeudi 23 février 2006 12:41 À : Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Objet : Re: [Asterisk-Users] IAX2 through Shorewall rpoblem I am trying to put a Shorewall firewall in front of my PBX, all the other port forwards work fine but forwarding port 4569 to the PBX is not working, it is being logged as rejected even though there is a DNAT rule in shorewall. Anyone seen this and have a solution? Are you sure its forwarding udp 4569 and not tcp? ___ --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] Detecting disconnect on TDM400P with 3 FXO portsand 1 FXS port
Hello Cosmin, This is extract from my zapata.conf : busydetect=yes busycount=3 busypattern=500,500 Check how is your local busy pattern for more efficiency. Good luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cosmin Prund Envoyé : mercredi 22 février 2006 11:16 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Detecting disconnect on TDM400P with 3 FXO portsand 1 FXS port Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to help Asterisk notice the disconect? This are the relevant parts of my zapata.conf: Callwaiting=no Usecallingpres=yes Callwaitingcallerid=yes Threewaycalling=no Transfer=yes Cancallforward=yes Callreturn=yes Echocancel=yes Echocancewhenbridged=no Echotraining=800 Rxgain=0.0 Txgain=0.0 Group=0 Callgroup=1 Pickupgroup=1 Faxdetect=incoming Immediate=yes Signaling=fxs_ks Context=from_rtc Busydetect=yes Channel = 4 ___ --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] Asterisk start errors with TDM2413E
Title: Message Hi, I believe that you have inverted fxo_ks and fxs_ks into your zapata.cong file "signaling=" declaration... Invert and redo the tests. Good Luck ! Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED]Envoyé: lundi 20 février 2006 04:34À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] Asterisk start errors with TDM2413EI get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Feb 19 21:14:35 ERROR[10440]: chan_zap.c:10264 setup_zap: Unable to register channel '1' Feb 19 21:14:35 WARNING[10440]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Feb 19 21:14:35 WARNING[10440]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# Ouch ... error while writing audio data: : Broken pipe Software versions asterisk-1.2.3 asterisk-addons-1.2.1 asterisk-perl-0.08 asterisk-sounds-1.2.1 libpri-1.2.2 zaptel-1.2.4 Output from modprobes [EMAIL PROTECTED] asterisk]# modprobe -v zaptel insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/zaptel.ko [EMAIL PROTECTED] asterisk]# modprobe -v wctdm24xxp install /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/wctdm24xxp.ko This takes at least 10 seconds to come back to a prompt ztcfg output [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) 16 channels configured. zaptel.conf fxoks=1-4 fxsks=5-16 defaultzone=us loadzone=us zapata.conf [channels] signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes usecallerid=yes context=outstation channel= 1-4 signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes usecallerid=yes group=2 context=incomingpstn channel= 5-16 Best regards,Duane PudenzNetwork Infrastructure ManagerShasta Industries ___ --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] Best quad-port fxo solution with EC?
Hi, I have good results with the new TDM2400P serie (with the hardware echocan, of course). May be you must check one TDM2401E to see if it's ok for you... Good luck. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : lundi 13 février 2006 07:36 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Best quad-port fxo solution with EC? Hello All, I am trying to figure out which way to go for a quad port fxo solution with a good echo can on it. My options are the sangoma remora, a mediatrix fxo, or something similar. The issue is that I would need a good EC. This would be on about a 9000 foot loop, and the lines don't function well on a spa-3000 or zaptel tdm 4 port card. Anyone have experience that drives them in a certain direction when considering a good ec on a quad port? I tried this also with some fxo clones, but echo killed it. Thanks, Greg ___ --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] To connect between more than 2 asterisk server [links needed ]
Hello, I have an IAX2 trunk like this running well with IAX2 and SIP users mixed at each side. Runing like a charm :-) Don't forget to add username definition from this example. To avoid too much load for your CPUs with transcoding, tempt to have only the same CODEC choice for all phones and participants. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de John Joseph Envoyé : dimanche 12 février 2006 16:07 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] To connect between more than 2 asterisk server [links needed ] Hi I am experimenting Asterisk , so far I am able to talk from two sip clients under one server and in the same network, [ Thanks to the mailing list ] Now I want to have two or more Asterisk server and SIP clients from one server communicating to the other sip clients in another server when I had searched I found this link http://www.voip-info.org/wiki-Asterisk+-+dual+servers I want to clear some doubts If I connect two servers using IAX , then will I have problems for SIP clients to communicate ? Is there any tutorials /notes/tips other than the above link for SIP/IAX connection between two servers ? Advice and guidance requested Thanks Joseph John ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.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 ___ --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] BAD/GOOD Echo Cancel
Hi the list, I can confirm you that I have not noticed any echo issue in this configuration (analog phones on quadFXS modules AND analog lines on quadFXO modules) at the same place and Asterisk when some echo issues occured with IP-Phones. TDM2400E is an excellent choice :-) Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de David Stude Envoyé : mardi 7 février 2006 17:09 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] BAD/GOOD Echo Cancel I've used Voicetronix FXO/FXS ports and noted pretty heavy echo on both short and long runs to other switches. We went through some steps to try to tune the echo out using some settings on the card, and it helped with some of the higher frequencies, but the problem still remains for many users. We decided, based on this and other problems, to pick up a Digium TDM board with 4 FXS ports and it virtually eliminated all our problems. The digium are short run (20 feet) to our PBX. The next step is probably going to be buying a 12 FXS / 8 FXO port TDM24XX card with hardware echo cancellation. The FXS will be all short run to our PBX and the FXO will be relatively long runs to the phone. So I'm very curious (and hopeful) that the problems will be much abated. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: Monday, February 06, 2006 5:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] BAD/GOOD Echo Cancel virtually all software echo cancelers cannot get double echo removed completly. It can get the first one but not the second one. There are instances where you get a 2nd echo, so ... Asterisk is no exception from this afaik nothing software only based is. If you really want good echo cancelation a hardware solution is the way to go. Just an enquiring mind wanting to know, but how is a hardware solution different to a software solution? The echo cancellers in the Digium hardware presumably just use the same sort of algorithms as the software versions, so it is just that they are dedicated and perform better, that they are closer to the source of the echo, or some other thing that I've overlooked? Thanks james ___ --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 ___ --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 : RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT
Argh ! Failed meeting with you ! Sorry ! Sure, Asterisk must be more present to this kind of exhibition. What could be the next french popular show at low price booth ? May be Mark could be there during it if we ask him ? Any idea of scenario or presentation ? -Message d'origine- De : Wilson Pickett [mailto:[EMAIL PROTECTED] Envoyé : vendredi 3 février 2006 09:52 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT Have you seen that 3 Asterisk servers were running during this show ? François, I was there (had a coffee with Dave in fact) but was wondering, there was no official asterisk presence, was there? Maybe we should have helped organize this as * is a Linux Solution ___ --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 : RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT
Interested, of course, but may be we can do that for a nearer exhibition this year ? Francois BERGERET. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Dave Cotton Envoyé : vendredi 3 février 2006 13:54 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote: Have you seen that 3 Asterisk servers were running during this show ? François, I was there (had a coffee with Dave in fact) but was wondering, there was no official asterisk presence, was there? Maybe we should have helped organize this as * is a Linux Solution Good idea, and we've got 362 days to organise it. I'd be ready to do it. It could be in the village or even a proper stand, what do the rest of the French users think? -- Dave Cotton [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] OT O'Reilly Asterisk TFOT
Hi Dave and the list, I was at this exhibition near all the first day. Next time, we must organise a meeting for handshaking and discussions for Asterisk lovers during a next exhibition ? Just by saying hello and what's happening to the list ? It could be cool to meet us in real world ;-) Have you seen that 3 Asterisk servers were running during this show ? Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Dave Cotton Envoyé : jeudi 2 février 2006 09:40 À : Asterisk List Objet : [Asterisk-Users] OT O'Reilly Asterisk TFOT I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton [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] How many digium cards per server ?
Hi Harry, How many IRQ do you have ? Be carefull for power supply is it is several TDM2460E (all FXS ports) ! It is better to use a seconf power supply... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : lundi 30 janvier 2006 10:00 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] How many digium cards per server ? Hello, How many digium cards is supported per asterisk server ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.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 ___ --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 TDM2400P's will a server take?
A PICMG card equiped with Pentium M CPU permits to reduce dragsticaly the power consumption. As this, you can use more power from your PSU for the interface cards. But, for several TDM2460E/B cards with a heavy traffic charge (many simultaneous rings), I believe that it could be better to use a second separate PSU for the cards. The peak consumption is about 120 W on the 12Vcc branch if all the 24 FXS are ringing together. I think that only an industrial PC can provide a so high power level without risk during years not a small office server. You must also think to add further fans to cool the box AND the FXS modules wich have small heatsinks for the hot components. Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rob Lith Envoyé : lundi 30 janvier 2006 19:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] How many TDM2400P's will a server take? And you should consider how many FXS's you're running. More than one card with all FXS's will require a turbo fan to cool and if they all ring you'll need a decent power supply to handle the power draw. Rob On 1/30/06, Steven Ringwald [EMAIL PROTECTED] wrote: Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will be able to service many more interrupts/card/channels than a 500 mHz Celeron. :-) 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 : SPAM: [Asterisk-Users] fxo/fxs cards with 8 ports
Buy a TDM2400P card with several quadFXO modules : 24 ports max :-) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de roswel ajf Envoyé : vendredi 27 janvier 2006 23:17 À : asterisk-users@lists.digium.com Objet : SPAM: [Asterisk-Users] fxo/fxs cards with 8 ports we have got asterisk 1.0 (over 1 yrs old) version and very old zaptel version. That code is working only with 8 or less ports (accumulative) on digium fxs/fxo cards (2 cards with 4 ports each). the questoin is, what if we want 12 ports?..well, really, i don't understand the limitations? is it simply zaptel driver code fix? or kernel fix? or technology limitation? donno any tips would help. we are though planning to move to latest asterisk 1.2.3 on linux 2.4. thanks, very much appreciate any comments. ___ --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] make linux26
Title: Message Hi Mike, You must continue - for zaptel only- to "make linux26", as it is described in the companion file "README.Linux26" in the Zaptel folder (/usr/src/zaptel). Read the text from this file, as suggested inits title: To build for Linux 2.6, first you must be sure that you have asymlink to your linux-2.6 sources in /usr/src/linux-2.6. The 2.6kernel no longer needs the full sourcecode to build against it. Youcan create the symlink to /lib/modules/`uname -r`/build/ and thenyou can type: # make linux26# make install Note that you will also need CRC-CCITT functions compiledwith your kernel or as a kernel module. These can beselected from the "Library Routines" submenu during kernelconfiguration via "make menuconfig" It is a good habit to read all this "README..." files before to do something, as it is important to read any user manual for any sofisticated equipment ;-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike HammettEnvoyé: lundi 23 janvier 2006 22:10À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] make linux26 I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.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] RJ21-RJ11
Buy the AMPHENOL 50pins male connector alone or with a pre soldered cable and do what you want with. Or buy a RJ11 pannel from the usual Telco accessories resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ing. Germán González B. Envoyé : vendredi 13 janvier 2006 23:56 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] RJ21-RJ11 Hi!! I'm looking for an adapter RJ21 to 24 RJ11 for a TDM2400. Somebody can help me with some sugestions? Thks!!! --- Germán González http://leon.podernet.com.mx --- ___ --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 : RE : RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found
Hi Chawki, I use a Debian Etch (testing branch) distro for my * box. Here, my zaptel modules are all in a zaptel folder, not in an extra. And the complete path owns the kernel name without any extension as yours. I am not sure of what to do... But, at your place, I will tempt two things : - copy your /lib/modules/2.6.8.1-12mdkcustom/extra (all the * concerned files) to a new folder /lib/modules/2.6.8.1-12mdksmp/zaptel and retempt to modprobe zaptel and all your necessary modules. - search if you have another folders branch from /lib/modules/ (tell us what you have here). Tell us what. Best Regards, Francois BERGERET, France. -Message d'origine- De : chawki hammoud [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 00:18 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found HI: It gives me this: Linux version 2.6.8.1-12mdksmp ([EMAIL PROTECTED]) (gcc version 3.4.1 (Mandrakelinux (Alpha 3.4.1-3mdk)) #1 SMP Fri Oct 1 11:24:45 CEST 2004 --- [EMAIL PROTECTED] wrote: What is the result of your cat /proc/version ? -Message d'origine- De : chawki hammoud [mailto:[EMAIL PROTECTED] Envoyé : vendredi 30 décembre 2005 23:21 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found Hi: I searched for zaptel.ko and i found it in lib/modules/2.6.8.1-12mdkcustom/extra ,is that the correct directory for zaptel.ko . ___ --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 __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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] name that vendor...
Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 ___ --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:problem with X100P card
Title: Message http://www.digium.com/index.php?menu=configuration RTFM ;-) Best Regards, Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tejas ShahEnvoyé: samedi 31 décembre 2005 06:04À: asteriskObjet: [Asterisk-Users] RE:problem with X100P card hi all, I wanted to knw whether it is possible to make call to analog phone (outbound call) using X100P card. I have only single piece of card. I m receiving call from analog phone properly,but cant make outbound call. If any one have a dialplan structure pls tell me.Thanks,Tejas Yahoo! for Good - Make a difference this year. ___ --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 : RE : [Asterisk-Users] name that vendor...
Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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 : RE : RE : [Asterisk-Users] name that vendor...
I believe that the Micronet firmwares authorize to have separate accounts for each different ports in SIP version. I will check this this next week at job and I will feedback you the results. Best Regards, Francois BERGERET, France. -Message d'origine- De : Vahan Yerkanian [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 14:04 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : RE : [Asterisk-Users] name that vendor... welltech... last time i tested their fxo 4 port gateway like year ago all ports were trying to communicate using same Call-ID. [EMAIL PROTECTED] wrote: Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --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 ___ --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 -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --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] Howto config tdm2400
Hello, Do as with a TDM400P, but use the correct driver (modprobe wctdm24xxp). You have only more channels, it's all ! Insert the quad modules starting from number 1 printed place on the PCB. This card run well and echocancel is very good. Good luck ! Francois BERGERET, [EMAIL PROTECTED], France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Manuel Casal Envoyé : vendredi 30 décembre 2005 13:06 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Howto config tdm2400 Hi, I've just received a brand new td2400e , Where i can found some documentation for this card?, Digium's site do not show very usefull. I'd like to know how to configure zaptel.conf and zapata.conf basically. Thanks, and Happy New Year to all. -- Manuel Casal [EMAIL PROTECTED] [EMAIL PROTECTED] Sistemas de Información y Protección de Datos, S.L. Telf. + 34 902 678006 e-mail: [EMAIL PROTECTED] web: http://www.e-sistemas.net ___ --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] Aterisk 1.2.1 zaptel module not found
Title: Message If yes, search if the modules are not in an any incorrect kernel branch if you have several : /lib/modules/2.6.12-1-686/zaptel/zaptel.ko May be it is in another branch as : /lib/modules/2.6.12-1-386/zaptel/zaptel.ko If yes, check your configuration (headers, kernel), recompile(the best way)or tempt to copy the modules in the correct branch. Good Luck ! Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Moises SilvaEnvoyé: vendredi 30 décembre 2005 22:33À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: Re: [Asterisk-Users] Aterisk 1.2.1 zaptel module not foundmm and sure you have compiled the zaptel packages and make install ? On 12/30/05, jonny hashem [EMAIL PROTECTED] wrote: Hi:i have compiled Asterisk 1.2.1 without any problems,But when i've tried to load the zaptel modules by making modprobe zaptel this message shown:FATAL: Module zaptel not found.Regards;jonny__Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection around http://mail.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-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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 : RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found
What is the result of your cat /proc/version ? -Message d'origine- De : chawki hammoud [mailto:[EMAIL PROTECTED] Envoyé : vendredi 30 décembre 2005 23:21 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found Hi: I searched for zaptel.ko and i found it in lib/modules/2.6.8.1-12mdkcustom/extra ,is that the correct directory for zaptel.ko . ___ --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] TDM2400
Hello folks ! TDM2400 with E for echocan module is ok for me, replacing my old passive cards. No more echo issues now. I had many before to switch to this wonderfull card ! Perfect for my use... Here is an Asterisk SVN-branch-1.2-r7608M, in an old PII-400 MHz Linux version 2.6.12-1-686 (gcc version 4.0.2 20050917 (prerelease) (Debian 4.0.1-8) Etch. My opinion : buy it WITH the echocan option, and don't forget to buy a Centronics 50 pins mâle connector (not provided). Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Guillermo Salas M Envoyé : jeudi 22 décembre 2005 17:01 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] TDM2400 Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] zapata directory not found in svn .
Hi Kevin and the list, Yes, please, you must. TIA Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Kevin P. Fleming Envoyé : mercredi 30 novembre 2005 15:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] zapata directory not found in svn . Tzafrir Cohen wrote: Is it obsoleted? It looks like a nice toy. See e.g. the recent http://linuxgazette.net/120/smith.html No, it's still on our CVS servers and will be there indefinitely. If there is demand (I assumed there wouldn't be) I can easily import it into SVN as well... ___ --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] In France asterisk never detect hang up. Why ?
Hello everybody :-) This are my first line french zapata.conf settings. I have 3 like this, with only rx/tx gain a little bit different levels. Running well. Best Regards, Francois BERGERET, France. usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=6 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=3 busypattern=500,500 signalling = fxs_ks channel = 1 -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de asterisk user dupont Envoyé : vendredi 18 novembre 2005 13:33 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] In France asterisk never detect hang up. Why ? Hello. I am sorry my english is not good at all. When i have a call from a fxo port of a tdm400p, asterisk waits one minute before detecting that the caller has hang up his phone. I have in my extension conf : answer background (the prompt is 40 second long) dial (on fxs port) confgured for 30 seconds ringing. if the caller hang up at the begining of the background prompt, asterisk waits until he make ring the phone on the dial command for the all 30 secondes before detecting the hang up. Do you know if there is a way to repair that ? here is what i see on asterisk when the caller hang up IMMEDITALY after the test prompt begins : *CLI -- Starting simple switch on 'Zap/4-1' -- Executing Answer(Zap/4-1, ) in new stack -- Executing NoOp(Zap/4-1, 0675458745) in new stack -- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack -- Response timeout set to 20 -- Executing BackGround(Zap/4-1, barge) in new stack -- Playing 'test' (language 'fr') -- Executing Dial(Zap/4-1, Zap/2|0675458745|30) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/4-1 -- Attempting native bridge of Zap/4-1 and Zap/2-1 -- Hungup 'Zap/2-1' == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1' -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' In my zapata.conf i have : language=fr default=fr relaxdtmf=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 cidsignalling=v23 usecallerid=yes group = 1 context=reseau signalling=fxs_ks callprogress=yes busydetect=yes callerid=asreceived busycount=5 pulse=yes In my zaptel.conf i have : loadzone=fr defaultzone=fr fxoks=1-3 fxsks=4 If anyone can see what is wrong he will really help me. thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Wits end with echo
1.2-beta2 is more efficient against echo issues with ECHO_CAN_MG2 :-) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jon Reynolds Envoyé : jeudi 10 novembre 2005 08:58 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Wits end with echo Richard Scobie wrote: Jon Reynolds wrote: I have updated the phones to 1.0.12 firmware, I have echotraining=800, echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using Mark2 as the echo suppresion and still I have echo. Is this correct? I do not believe having these echo parameters in sip.conf will achieve anything. They should be at the top of zapata.conf. Regards, Richard That is incorrect, I wasn't thinking clearly, it is zapata.conf that these settings are in. Thanks for the correction Richard, Jon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)
Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. The first error message happens by using the famous script from http://www.szmidt.org/asterisk/asterisk-update.sh : configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 ERROR! Compile exited with error. Aborting script! And, if I tempt to compile manualy with make clean; make; make install, I can see that at the end : cd editline unset CFLAGS LIBS test -f config.h || ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i686-pc-linux-gnu cygwin detected checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 sarge:/usr/src/asterisk# What occurs ? What I have missed ? Any idea to help me ? What can I describe or search more for a best analyze ? Many thanks in advance, guys ! Best Regards, Francois BERGERET, France. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] IAX2 hard phone
Hello Alberto, You must upgrade the firmware by taking the last one at www.aredfox.com which is the PA168 manufacturer. Mine Ip-phones are running well with IAX2 and flash hook for transferts. Good luck. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Alberto Risco Envoyé : mardi 27 septembre 2005 15:06 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] IAX2 hard phone I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays transferring but it does nothing. Has anybody with these phones run into similar problems? Or can recommend a good functional IAX2 hard phone. Thanks, Alberto The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)
Many thanks Tzafrir and Sergio, Now, I have another error when compiling zaptel : /lib/modules/2.6.8-2-686/build make -C /lib/modules/2.6.8-2-686/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernel-headers-2.6.8-2-686' CC [M] /usr/src/zaptel/zaptel.o In file included from include/asm/thread_info.h:16, from include/linux/thread_info.h:21, from include/linux/spinlock.h:12, from include/linux/capability.h:45, from include/linux/sched.h:7, from include/linux/module.h:10, from /usr/src/zaptel/zaptel.c:44: include/asm/processor.h:87: error: array type has incomplete element type /usr/src/zaptel/zaptel.c: In function '__zt_receive_chunk': /usr/src/zaptel/zaptel.c:6115: warning: pointer targets in assignment differ in signedness make[2]: *** [/usr/src/zaptel/zaptel.o] Erreur 1 make[1]: *** [_module_/usr/src/zaptel] Erreur 2 make[1]: Leaving directory `/usr/src/kernel-headers-2.6.8-2-686' make: *** [linux26] Erreur 2 sarge:/usr/src/zaptel# What to do more ? -Message d'origine- De : Sergio Serrano [mailto:[EMAIL PROTECTED] Envoyé : mardi 27 septembre 2005 09:36 À : [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1) You must install libncurses5-dev regards, Srsergio -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tzafrir Cohen Envoyé : mardi 27 septembre 2005 09:33 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1) On Tue, Sep 27, 2005 at 09:20:13AM +0200, [EMAIL PROTECTED] wrote: Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. Start with defining a standard deb-src of Sarge (I think it is defined by default. Maybe remmed-out) and then run: apt-get install build-essential apt-get build-dep asterisk It should get you roughly the packages needed to build HEAD from source. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] civil emergency comms: Asterisk + HAM
Hello, Asterisk is on the air : http://www.hamwlan.net http://192.168.1.1/HamWlan.htm (see the second drawing) 73 ! F6HQZ, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Hemstock Envoyé : jeudi 8 septembre 2005 21:29 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM On Tuesday 06 September 2005 15:27, Mike M wrote: The disaster in the Gulf coast and the less than optimal initial response suggests to me that citizens must shoulder more responsibility for emergency management. Communications loss must have played a large role in the failures that occurred. I can't help but wonder if there are fewer ham radio operators today and that if there were more, maybe they could make a difference in future emergency situations. Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. I suspect there are some folks on this list that have already implemented such a system. If so, I would like to read about what they have done so I can develop a plan to participate in this network if one exists. There's not much on google for asterisk ham. http://sourceforge.net/projects/hamlib/ http://www.radioadv.com/default.htm Thanks, That's a very interesting idea. I believe radio ametures who have a radio in their car don't have to pay road tax in Canada as they one provided emergency comms during a civil emergency. Mike. 2E1HFW ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Motherboard and processor recommendations
Hello Men, And what about of industrie PC's with passive PCI slots buses ? You can upgrade it easily by changing its daughter card supporting the CPU and main chipsets instead of changing a complete motherboard? Power supplies are often bigger/stronger than standard tower PC. My 2 cents. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Remco Barende Envoyé : vendredi 9 septembre 2005 23:40 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Motherboard and processor recommendations I'm looking for a good, reliable and upgradeable solution too. I don't care to spend a lot of money if the hardware is reusable. A Dell 2850 is useless after 3 years, no way to upgrade it. A quality Intel SC5300 for example is not cheap at all but will last you a lifetime. ...SNIP... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [Asterisk-Users] civil emergency comms: Asterisk + HAM
Oops ! Sorry ! It seems that I have forgotten to replace my french characters as é by the correct sequence eacute; as exemple. I have just modify this page and you can probably read it now (but it is in french only for now, I promess to translate this pages this next cold season). HamWlan is now an old project from about 3 years old. This was to preserve our sub-band shared on the 2.4 GHz by ISM band and UHF HAM at WiFi market operture. This could be a nice opportunity to test high speed radio or HAM services as we have never seen until now (on HAM bands). It is also to use our 44.x.x.x/8 IP addresses class reserved to our HAM community. I have started some VPN under IPSec to separate public traffic from HAM's traffic as lawyers said in near all the countries. This HAM's hotspots are connected as this through Internet if not possible by radio link. To attract HAMs to join this fun wireless project, I have added some classical services encountered on Internet : SMTP/POP, H.323 video conference and Jabber servers. I have also started an IP gateway between HF 7 MHz band and my IP local network. As I am self training on Asterisk from monthes and use one at my home for my own private telephone lines, I have think that it could be nice to connect my Asterisk box to my HamWlan network (without any telephone access because it is forbidden in France). I am just starting to tell to some HAMs to join me and start some experimentations to see if Asterisk could be interesting for HAM use. HAMs are already using some kind of Internet VoIP as Skype or Echolink are (Echolink is a HAM network connecting people and radio equipments). With Asterisk, we can use conference rooms (mine is [EMAIL PROTECTED]) or to share an UHF repeater linked to a room or a specific number. I have not enougth bandwith as I desire... I have two providers and the best is about 2.6 Mbs download and 650 kbs upload. The ideal way could be to place an asterisk in a ITSP white room with bigger bandwidth, but it is a dream only :-) For now, it is only the beginning, and I play to see if any HAM's interest. Best Regards, 73's from F6HQZ, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike M Envoyé : samedi 10 septembre 2005 22:22 À : asterisk-users@lists.digium.com Objet : Re: RE : [Asterisk-Users] civil emergency comms: Asterisk + HAM On Sat, Sep 10, 2005 at 09:08:53AM +0200, [EMAIL PROTECTED] wrote: Hello, Asterisk is on the air : http://www.hamwlan.net http://192.168.1.1/HamWlan.htm (see the second drawing) 73 ! F6HQZ, Francois BERGERET, France. Excellent. So you have SIP/IAX clients connecting to a router over HAM radio links, and the router is on a WLAN with an Asterisk box ( 44.151.177.66 : serveur Asterisk (PBX VoIP : SIP/H.323/IAX))? What sort of bandwidth is available on the hamwlan? I tried several different character encoding choices and I just couldn't get the proper representaions for the characters on the web page. Can you recommend an appropriate character set for Firefox for French? Babelfish will probably work better if I used the correct character set. Thanks, -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. Dear JR, I am realy sorry about all this desaster, but happy to see you alive. For waterlogged equipments, no problem until they are under the water level (out of oxygen contact). The best way after that will be to clean all of this waste and poluted equipments with clear water during a long time (no more chimical products in the clean water), and to quickly dry them. Good luck. Best Regards to you and your family. Francois BERGERET, France. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] How to shorten ringing stop detection onX101Pclone?
Hello Goran, Yes, you are right ! I have read too quickly your question, sorry. I have checked myself with my * and same thing appears here. If you will have the solution before me, please, post it :-) For F.T. prices list, it is not realy easy... You can start by browsing from here : http://www.agence.francetelecom.com/vf/home_pro/index.htm And here : http://www.francetelecom.com/fr/entreprises/grandes_entreprises/solutions/re seaux/tarifs_voix/plan_tarif/att00028453/tarifs.html But, there is a lot of different contracts and discounts... Good Luck. Best Regards, Francois BERGERET, France. -Message d'origine- De : Goran Dj. [mailto:[EMAIL PROTECTED] Envoyé : jeudi 1 septembre 2005 22:48 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] How to shorten ringing stop detection onX101Pclone? This working only when zap answer call. But, if zap don't answer (ringing), and (outside) caller hangup, then there is no busy tone. By the way, do you know some voip provider in Paris with Direct Inward Dial numbers? Where can I found best information about prices of France Telecom (PRI od BRI ISDN/RNIS, tarrifs, etc...) - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: cet 1. sep 2005 7:13 Subject: RE : [Asterisk-Users] How to shorten ringing stop detection onX101Pclone? Hello Goran, Modify your /etc/asterisk/zapata.conf like this : busydetect=yes busycount=3 And, of course, you must have chosen your correct country for ringing mode in your /etc/zaptel.conf file : loadzone=fr defaultzone=fr I am in France :-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Goran Dj. Envoyé : jeudi 1 septembre 2005 02:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] How to shorten ringing stop detection on X101Pclone? When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] How to shorten ringing stop detection on X101Pclone?
Hello Goran, Modify your /etc/asterisk/zapata.conf like this : busydetect=yes busycount=3 And, of course, you must have chosen your correct country for ringing mode in your /etc/zaptel.conf file : loadzone=fr defaultzone=fr I am in France :-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Goran Dj. Envoyé : jeudi 1 septembre 2005 02:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] How to shorten ringing stop detection on X101Pclone? When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Dial Zero to get outside line?
Hello Michael and the list,
exten = 0,1,Dial(Zap/3/${EXTEN:1}) ; call to RTC FXO#3, digit #1 (0)
suppressed
I hope this could help some.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Michael
Felder
Envoyé : mercredi 24 août 2005 07:04
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] Dial Zero to get outside line?
Hello Craig,
Yes I would like to dial 0 to get an outside line and dial tone, then dial
the number.
I have Polycom IP600 and IP 500s.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Monday, 22 August 2005 10:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial Zero to get outside line?
Hi Michael,
What phones are you using as this will affect your implementation. For
example do you want to dial zero, then hear a dialtone and dial the full
number or do you wish to dial the whole number with a preceeding zero in one
hit?
Craig
- Original Message -
From: Michael Felder [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 22, 2005 9:17 AM
Subject: [Asterisk-Users] Dial Zero to get outside line?
Hello,
My asterisk currently will dial an outside number after I dial the number
and press send on the phone. How can I get it setup so I have to press '0'
for an outside line.
Kind regards
Michael Felder
IT Medic Australia Pty. Ltd.
P: 03 9557 2213
F: 03 9557 2214
M: 0419 568 217
E: [EMAIL PROTECTED]
http://www.ITMedic.com.au
Keeping your computer systems healthy.
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[Asterisk-Users] French national telco 1004hz test phone number ?
Hello Asterisk friends, Does somebody know few french phone numbers to do telco 1004Hz 0dBm signal tests phone ? Thanks in advance. Best Regards, Francois BERGERET, Happy French Asterisk user :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [Asterisk-Users] How to read dbm or voltage via ztmonitor ?
Thank you for our point of vue, Rich :-) Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rich Adamson Envoyé : mardi 5 juillet 2005 14:40 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] How to read dbm or voltage via ztmonitor ? Yes, I've read that. Ztmonitor is simply a very _basic_ tool that provides you with a little bit of feedback to adjust the rxgain and txgain settings to something relatively close to what the human ear considers reasonable audio level. ...SNIP... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
