[Asterisk-Users] Call transfer.
Hello, how i can tranfer call to another user? Im using X-Lite, i have configured in features.conf: [featuremap] blindxfer = #1 disconnect = *0 automon = *1 atxfer = *2 But when im dial *2 in conversation nothig happens. What can br problem? Im using asterisk CVS-HEAD from 02/09/05. Regards, Adam Rybak ___ --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] DID on analog line
Peder @ NetworkOblivion wrote: And it's wink-start on an EM analog circuit, not on a standard analog phone line from your telco. You would need a card that supports EM to do it even if the telco provided it (not sure if the Digium cards support it, but I tend to doubt it). We do not have any four-wire analog cards, so we cannot handle analog EM signaling. We do support EM over digital links, though. Analog DID can be done using ground start as well, so it's possible than an FXS port could be convinced to do it, but nobody has implemented it, since analog DID is not something that carriers really want to continue selling. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Marco Balmer wrote: Any ideas or hints? Yes. Whatever documentation told you that you could share a Realtime SIP peer database between two Asterisk servers was in error (or at least very incomplete). There are ways to do it right now, but it's not trivial and does not provide all the functionality that someone would want from such an arrangement. ___ --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] Sound too loud (saturated). How to change?
I have very loud sound through IAX2 channel,very saturated in some moments.How to find where is problem. I think problem is at provider side, but how to be doubtless? Is there any method to measure and change sound level on IAX channel (like on Zap channel)? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
I think, that mistake is between PC and chairs. When i have not outgoing lines it's too hard to call out. Now i'm in state, that example form README dialed and i'm trying to receive fax on other side. Thanks, Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Thursday, October 13, 2005 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Yeah I missed that in the original, sorry bout that. are you sure that the other end didnt hang up? You may want to test this by calling a number you have access to so that you can at least rule that out. The only other thing I can think of is that txfax itself is aborting and returning prematurely. I wonder if its a negotiation failure. You say it hangs up immediatly, how immediatly? 1 second? 5? On Thu, 2005-10-13 at 11:52 +0200, Coufal Bohuslav wrote: But it seems that Asterisk understand that he has to dial (the dialed number is correct), -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) it seems that zap channel had answered (but nothing to try dial), Channel Zap/4-1 was answered. and lunching txfax Launching txfax(/tmp/ast_fax-1129191936.10240.1804289383.0|caller) on Zap/4-1 and immediately hungup -- Hungup 'Zap/4-1' May be something wrong in zapata.conf? ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] ; language=us signalling=fxs_ks context=default ;context=fax channel = 3-4 Thank for any other sugestions, Bob. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Hello On Fri, 14 Oct 2005 01:25:20 -0500, Kevin P. Fleming wrote Marco Balmer wrote: Any ideas or hints? Yes. Whatever documentation told you that you could share a Realtime SIP peer database between two Asterisk servers was in error (or at least very incomplete). Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. Thanks Marco ___ --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] DID on analog line
Multitech makes ATAs and Gateways that support EM signaling: http://www.voip-info.org/wiki/view/Multitech - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 13, 2005 8:23 PM Subject: Re: [Asterisk-Users] DID on analog line Peder @ NetworkOblivion wrote: And it's wink-start on an EM analog circuit, not on a standard analog phone line from your telco. You would need a card that supports EM to do it even if the telco provided it (not sure if the Digium cards support it, but I tend to doubt it). We do not have any four-wire analog cards, so we cannot handle analog EM signaling. We do support EM over digital links, though. Analog DID can be done using ground start as well, so it's possible than an FXS port could be convinced to do it, but nobody has implemented it, since analog DID is not something that carriers really want to continue selling. ___ --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] sip accounts
hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info [EMAIL PROTECTED] disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus anybody can just use my system to call others.. lets say i do have set that some certain account can make some certain calls only. how can i solve this problem? thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy build problems
On Thu, 2005-10-13 at 17:02 -0700, Bruce Ferrell wrote: Hi all, Trying to build ztdummy on an old redhat 7.3 box running kernel 2.4.20-43.7.legacysmp. Yes, I have the kernel sources installed. Yes, I set them up with make oldconfig; make dep. The build error is: make ztdummy gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/ linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPAT A -c ztdummy.c cc ztdummy.o -o ztdummy /usr/lib/gcc-lib/i386-redhat-linux/2.96/../../../crt1.o: In function `_start': /usr/lib/gcc-lib/i386-redhat-linux/2.96/../../../crt1.o(.text+0x18): undefined reference to `main' ztdummy.o: In function `init_module': ztdummy.o(.text+0x7): undefined reference to `uhci_devices' The above is pointing to USB devices are they configured in the kernel and does the machine support uhci? Suggestions? 2.6 ztdummy does not need USB devices. gcc 2.96 don't know if that also will have probs long time since I used it. Upgrade it's free. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?
Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect Polarity Reversal:yes Detect Disconnect Tone: yes, with the default value. Thanks a lot for your help ;) ___ --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] [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
Title: Patrick Briefpapier
All,
Currently I've got
my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones
or softphones within my LAN or remote LAN's via VPN. The next step for me is
connecting it to the PSTN network.
After some tweaking
with the modem.conf I got the i4l driver running correctly, and it appears that
my Fritz! ISDN v2 card is working correctly.
I have added the
following to my extension.conf file:
[extensions.conf]
exten =
200,1,Dial(Modem/ttyI0:00104431040)exten =
201,1,Dial(Modem/ttyI1:00104431040)exten =
202,1,Dial(Modem/ttyI2:00104431040)
exten =
203,1,Dial(Modem/g1:00104431040)
When I dial either
extension it shows the following error:
-- Executing
Dial("SIP/102-36a5", "Modem/g1:00104431040") in new stack
-- Called g1:00104431040Oct 14 09:42:41 WARNING[2901]: chan_modem_i4l.c:380
i4l_read: Device '/dev/ttyI1' lacking dialtone -- Hungup
'Modem[i4l]/ttyI1' == No one is available to answer at this time
(1:0/0/0) == Auto fallthrough, channel 'SIP/102-36a5' status is
'NOANSWER'
I have found
some references in previous mailings, but none of them seem to solve the problem
in my situation.
I hope someone
here can help me!
Thanks in
advance,
Patrick
---
This email was scanned by MyMail of DatacomPartner
http://www.datacompartner.com
___
--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] sip accounts
On 10/14/05 15:42 Kong said the following: but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus anybody can just use my under the [general] section, use a context which limits what unauthenticated users can do/call. it can even be a catchall IVR which says bugger off !. :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
On Fri, 2005-10-14 at 10:15 +0200, Patrick de Kok wrote: All, Currently I've got my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones or softphones within my LAN or remote LAN's via VPN. The next step for me is connecting it to the PSTN network. After some tweaking with the modem.conf I got the i4l driver running correctly, and it appears that my Fritz! ISDN v2 card is working correctly. I have never used i4l with * but have systems running without any problems using chan_capi and 1 or even 2 Fritz! cards. The only 'tweaking' needed was the patching for the second card. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip accounts
how to chech if the user is an unauthenticated one? thank you At 03:58 PM 10/14/2005, you wrote: On 10/14/05 15:42 Kong said the following: but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus anybody can just use my under the [general] section, use a context which limits what unauthenticated users can do/call. it can even be a catchall IVR which says bugger off !. :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Reset telephone IP PHONE 106
I have a telephone Voismart IP PHONE 106. I have lost the password of the telephone and therefore I am not able to set up it. How can I do to do a reset of the telephone? ___ Yahoo! Messenger: chiamate gratuite in tutto il mondo http://it.messenger.yahoo.com ___ --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] Access to trunks
Are there any configuration options to allow certain sip/iax accounts to dial out over specific trunks, and also to stop them dialing out over other trunks. Thanks in advance Bails ___ --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] DID on analog line
On Thursday 13 October 2005 15:20, Apu Islam wrote: Is DID on analog line possible ? ( my telco is qwest) . Just wondering if there is any way to test it on anlog wcfxo cards. Another approach is to use a CTPX or Exacom unit to convert the DID or 2-Wire EM signal into a signal appropriate for a Digium FXO port (converting the DID signaling info into DTMF after Asterisk takes the FXO port off-hook). See: http://www.ctpx.com/html/vp_2000__product_series.html and http://www.exacom.com/html/vertical_markets/did_solutions.htm -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks
On Mon, 2005-10-03 at 17:54 -0400, Matt Roth wrote: List members, 2) What will happen on the NFS client if the NFS server crashes (I expect the leg files to be written to the local mount point until the mount is reesablished)? Why don't you create a file on the NFS server called something like nfsmount then your script which copies files from RAMdisk to NFS would check for the existence of a file called nfsmount, if it isn't there you either copy the file to a local partition, or delay to copy until the NFS mount returns or copy to a different NFS server (backup) etc... Regards, Adam ___ --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] sip accounts
On 10/14/05 16:40 Kong said the following: how to chech if the user is an unauthenticated one? thank you read www.voip-info.org on SIP. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Which H323 module to go for?
I want to add H323 support to my asterisk setup. What are the pros and cons of the available modules, h323, oh323 and ooh323 and which is the best one to go for? Obelix This message was sent using IMP, the Internet Messaging Program. ___ --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] [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
Title: Patrick Briefpapier Some additional information: mchan_modem.so[0;37;40m] = ([33;40mGeneric Voice Modem Driver[0;37;40m)Parsing '/etc/asterisk/modem.conf': FoundLoading modem driver chan_modem_i4l.so = ([33;40mISDN4Linux Emulated Modem Driver[0;37;40m)Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)Configured modem /dev/ttyI1 with driver i4l (Linux ISDN)WARNING mchan_modem.c mload_module Unknown dtmf detection mode 'asterisk/both', using 'asterisk'WARNING[mchan_modem.cmload_module Unknown dtmf generation mode 'asterisk/both', using 'asterisk' This is the only warning Asterisk gives when starting up. Thanks for any help! CONFIDENTIALITY NOTICE: This message and its contents are confidential and are intended solely for the use of the addressee. If you are not the intended recipient, you are hereby notified that any use, dissemination, distribution or copying of this message and its contents is strictly prohibited. In addition, please contact the sender by email and delete the original message immediately. Thank you for your cooperation. --- This email was scanned by MyMail of DatacomPartner http://www.datacompartner.com ___ --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] sip accounts
can i know where to start? SIP is such a big topic. At 05:58 PM 10/14/2005, you wrote: On 10/14/05 16:40 Kong said the following: how to chech if the user is an unauthenticated one? thank you read www.voip-info.org on SIP. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] three way calling
hi can i make sip three way call on asterisk i meen one person call one time to two another and when they answer this 3 person speak with each other as in confereces i cant use meetme becouse i need send dtmf -- Oleh Mukha IClub 380322722738 www.ic.lviv.ua ___ --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] Incoming call problem - ringing SIP devices report busy
Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so: Dial (SIP/1SIP/2etc.) This has worked fine for some months, but I noticed a few days ago that if 2 calls come in only a second or two apart, the first one will cause the dial command to be executed, and when the second call comes in, it'll go to voicemail because *all* the SIP phones report themselves as busy (because they're ringing for the first call). Is there any way around this problem whilst keeping the same incoming call behaviour (i.e. call comes in, all phones ring)? It's vitally important that asterisk does *not* answer the incoming call until the call has actually been connected to a SIP phone (or voicemail). Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] sip register incoming call contexts?
Hello,
I have set up 2 different fwd.pulver.com accounts on my Asterisk. One will ring all my phones through one context, while the other account was set up to fool Nigerian scam artists, and will go directly to a special voicemail (after a few rings to give the impression of ringing a real telephone). I will not even know somebody called until I get the voicemail in the mail.
The first register goes like this:
register = 18469:[EMAIL PROTECTED]/89
while the number that goes directly to the answering machine is as follows:
register = 18336:[EMAIL PROTECTED]/36
Then I match the digits (36 and 89)within the contexts.
89 triggers the [inbound-fwd] context, while 36 triggers [boguscall]:
[boguscall]exten = 36,1,NoOp(This is context boguscall)exten = 36,2,Wait(0)exten = 36,3,Ringingexten = 36,4,Wait(15)exten = 36,5,Voicemail(su36)exten = 36,6,Hangup
[inbound-fwd]
exten = 89,2,Goto(ringall,${EXTEN},1) ; will go to context [ringall]
[ringall] ; Dial all telephones in the houseexten = _X.,1,Dial(SIP/30SIP/31SIP/32,35),t
Thor
On 10/10/05, Steve Gladden [EMAIL PROTECTED] wrote:
Sorry this is a bit of a newbie question, I've been at this for a fewmonths and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with aregister line like this:register = nnn:[EMAIL PROTECTED]
-or-register = nnn:[EMAIL PROTECTED]/nnnto come directly into an extension in the dialplanIt seems that this only works with the default context in the dialplan.
I have another sip account from another provider that I would likeall of it's incoming calls to come into the s, extension ofa new context but I have been unable to figure outhow to bring calls from a register line into an alternate context.
It seems that register lines are limited to only being used in thegeneral section of sip.conf and you are limited to one context=statement there.Is there a way to register a second account and have it's calls come into
another context in the dialplan?register lines only seem to work in [general] and it seems like youare limited to only one inbound context here.I would like the two inbound call accounts to be 'isolated' from each other
and not have to come in on the same incoming context in the dialplan.I'd also like to be able to have them have their own contexts with thierown s, (start) extension available.Thanks!Steve
___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://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] Asterisk IAX config user
Hello,I am trying toconfig inter Asterisk IAX2 connection. When I register a username and password it works but I would like that "Any" incomming SIP call (without specific username and password) pass throught IAX2 for delivery to the other end *.Is it possible ?I read in Asterisk IAX config, if no username is specified at all, Asterisk will authenticate the connection as the guest user but when I try it does not work eventhough I have guest declared in iax.conf: [guest] type=user context=guest Also,regarding optimization it seems that I can not use trunk=yes since I need a digium card in each pbx for timing. Any suggestions, hint would be greatly appreciated, Thanks,Frank Yahoo! Music Unlimited - Access over 1 million songs. Try it free.___ --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] SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?
makevuy wrote: Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect Polarity Reversal:yes Detect Disconnect Tone: yes, with the default value. Thanks a lot for your help ;) I've never used one of these (but I'd like one). However, if it is not detecting the disconnect tone, it could be that your telephone service provider is providing a tone is not the same as the one the unit is expecting. For example in the UK you need to change the settings for the disconnect tone from the defaults. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
All works very well. Last question is if there is a chance to get result of sending by mail (for example as answer to my mail). Thanks, Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Thursday, October 13, 2005 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Yeah I missed that in the original, sorry bout that. are you sure that the other end didnt hang up? You may want to test this by calling a number you have access to so that you can at least rule that out. The only other thing I can think of is that txfax itself is aborting and returning prematurely. I wonder if its a negotiation failure. You say it hangs up immediatly, how immediatly? 1 second? 5? On Thu, 2005-10-13 at 11:52 +0200, Coufal Bohuslav wrote: But it seems that Asterisk understand that he has to dial (the dialed number is correct), -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) it seems that zap channel had answered (but nothing to try dial), Channel Zap/4-1 was answered. and lunching txfax Launching txfax(/tmp/ast_fax-1129191936.10240.1804289383.0|caller) on Zap/4-1 and immediately hungup -- Hungup 'Zap/4-1' May be something wrong in zapata.conf? ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] ; language=us signalling=fxs_ks context=default ;context=fax channel = 3-4 Thank for any other sugestions, Bob. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --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] Calibrating both RX and TX gain?
[EMAIL PROTECTED] wrote on 10/12/2005 01:23:57 PM: On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote: I am in the middle of trying to get a milliwatt test line to calibrate the rxgain properly. However, this won't help me with the txgain, will it? How can I properly calibrate the txgain? By ear? Or is there a more scientific method? I contacted Rhino to see if they had any suggestions, and they were able to give me a few. What finally worked was setting the Asterisk gains back to 0 for all channels, then adjusting the gains down on the channel banks themselves for the phone (FXS) interfaces only. A huge improvement! My current adjustements are the following: According to the company that installed the channel bank, there is a 0db and -10db setting on the smart jack for the T1. They claim that this was most likely set to -10db by the ILEC when the T1 was installed, and that would be causing the low audio volume. Does this make sense to anyone? Wouldn't the -10db affect the *digital* levels, not the analog waveform encoded within the digital signal? I'm still trying to get a milliwatt test line to calibrate from. They claim that they won't give that out to end users because it could fry the T1 card. Sigh. Tim Massey ___ --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] Music on hold disappears for Dial(, m) when calling outside numbers
Thanks. I'm prevented from testing it right now, but I will as soon as possible. It seems to be the fix that I need. Lars. On 10/13/05, Matt [EMAIL PROTECTED] wrote: Try disabling inband call progress tones. Let Asterisk handle everything. In sip.conf add the line: progressinband=no On 10/13/05, Lars Dybdahl [EMAIL PROTECTED] wrote: My asterisk is purely connected to the outside world via SIP. When I use Dial() with the m-option, that should ensure music-on-hold, it works perfectly as long as I am calling a SIP number, but when I call a mobile phone, the music-on-hold disappears. Any ideas on the cause of this or how to fix this? ___ --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] Problem with two hfc-s cards
Hi! I have installed two hfc-s cards to handle my pstn calls. I use mISDN with capi, so capi.conf is edited. I have tested both separate and cards are working well. But they are not working together. It seems that when i set up settings for the other card: ;capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=50 incomingmsn=* controller=1 softdtmf=0 context=demo devices=2 msn=50 incomingmsn=* controller=2 softdtmf=0 context=demo devices=2 only controller 1 is working. capiinfo shows me both cards properly. when i tried this capi.conf configuration: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=50 incomingmsn=* controller=1,2 softdtmf=0 context=demo devices=2 the situation is the same. Is there some thing I miss here? I really appreciate your answers. This mail sent through L-secure: http://www.l-secure.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reset telephone IP PHONE 106
The phone carries their configuration from the TFTP server, regarding the manufacturer. You should be able to change the password from the configuration file on the TFTP server. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabio Montemaggiore Sent: Friday, October 14, 2005 5:15 AM To: asterisk Subject: [Asterisk-Users] Reset telephone IP PHONE 106 I have a telephone Voismart IP PHONE 106. I have lost the password of the telephone and therefore I am not able to set up it. How can I do to do a reset of the telephone? ___ Yahoo! Messenger: chiamate gratuite in tutto il mondo http://it.messenger.yahoo.com ___ --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] T1/E1 Cards
This is probably a really bad question to ask but here goes. Does asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is a competitor to Digium, but I have access to SBE cards and the linux driver. Just curious more than anything. Thanks. -- Michael J. Lynch What if the hokey pokey IS what it's all about -- author unknown ___ --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] SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?
Where can I find this information? Faris Raouf wrote: makevuy wrote: Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect Polarity Reversal:yes Detect Disconnect Tone: yes, with the default value. Thanks a lot for your help ;) I've never used one of these (but I'd like one). However, if it is not detecting the disconnect tone, it could be that your telephone service provider is providing a tone is not the same as the one the unit is expecting. For example in the UK you need to change the settings for the disconnect tone from the defaults. Faris. ___ --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] Access to trunks
Bails wrote: - Are there any configuration options to allow certain sip/iax accounts to dial out over specific trunks, and also to stop them dialing out over other trunks. Thanks in advance Bails = Bails, Set the extensions to use certain context's. Example: - 1234, 1235, 1236 use context1 which dials out on ZAP/1 1237, 1238, 1239 use context2 which dials out on ZAP/2 etc. Dave ___ --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] [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
Title: Patrick Briefpapier I would prefer to get it working with i4l at the moment, and migrating later on to CAPI if needed. Thanks for any help you can give me.. - Patrick --- This email was scanned by MyMail of DatacomPartner http://www.datacompartner.com ___ --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] DID on analog line
George Pajari wrote: On Thursday 13 October 2005 15:20, Apu Islam wrote: Is DID on analog line possible ? ( my telco is qwest) . Just wondering if there is any way to test it on anlog wcfxo cards. Another approach is to use a CTPX or Exacom unit to convert the DID or 2-Wire EM signal into a signal appropriate for a Digium FXO port (converting the DID signaling info into DTMF after Asterisk takes the FXO port off-hook). See: http://www.ctpx.com/html/vp_2000__product_series.html and http://www.exacom.com/html/vertical_markets/did_solutions.htm I used such convertors about 15 yrs ago with a dialogic D41 4 port ISA card. For software testing I used a panasonic 6x16 switch without the convertors. The way I would test DID handling is that I relaxed the timing during the test. I would dial an extension connected to the dialogic card and soon as it answered send 4 digits. To make this easier I got a phone with a lot of autodialer buttons. Since Apu mentioned this is needed for a test I wonder if he can do that with DID's provided via SIP/IAX2? Last time I checked analog DID trunks were expensive both NRC and MRC. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Marco Balmer wrote: Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. But this is not currently implemented. There is a patch in the bug tracker that will help move in this direction, but it's only a start, there are many more issues that need to be resolved for this to work properly. ___ --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] T1/E1 Cards
I asked the same to Ben Dewey (SBE) a couple of weeks ago, and I get no answer. As I have a couple of cards, and I know that I can do channelized with those card, I believe that all that I should do is try it. If you know something different, let us know. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Lynch Sent: Friday, October 14, 2005 8:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] T1/E1 Cards This is probably a really bad question to ask but here goes. Does asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is a competitor to Digium, but I have access to SBE cards and the linux driver. Just curious more than anything. Thanks. -- Michael J. Lynch What if the hokey pokey IS what it's all about -- author unknown ___ --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] Problem with two hfc-s cards
What do you mean with 'not working'? Do you get any error messages? What does the log show? Do both cards work without asterisk/chan_capi? Armin On Fri, 14 Oct 2005 [EMAIL PROTECTED] wrote: Hi! I have installed two hfc-s cards to handle my pstn calls. I use mISDN with capi, so capi.conf is edited. I have tested both separate and cards are working well. But they are not working together. It seems that when i set up settings for the other card: ;capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=50 incomingmsn=* controller=1 softdtmf=0 context=demo devices=2 msn=50 incomingmsn=* controller=2 softdtmf=0 context=demo devices=2 only controller 1 is working. capiinfo shows me both cards properly. when i tried this capi.conf configuration: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=50 incomingmsn=* controller=1,2 softdtmf=0 context=demo devices=2 the situation is the same. Is there some thing I miss here? I really appreciate your answers. This mail sent through L-secure: http://www.l-secure.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1/E1 Cards
Michael J. Lynch wrote: This is probably a really bad question to ask but here goes. Does asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is a competitor to Digium, but I have access to SBE cards and the linux driver. Just curious more than anything. Thanks. SBE does not currently provide any support for their cards to be used with Zaptel. ___ --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] Don't know what to do if second ROSE component is of type 0x6
Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! Tom ___ --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] Voicemail - new feature request
Hi, I don't if was yet an issue. It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Regard ___ --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] 488 Not acceptable here
I have been receiving a lot these 488 Not Acceptable Here from a number of providers. What could the problem be? What is the most common cause of this message? This message was sent using IMP, the Internet Messaging Program. ___ --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] DTMF tones not working with SIP
My Asterisk PBX seems unable to receive DTMF information via SIP. I have tried all the various methods, rfc2833, inband and info and they all don't seem to work. IAX2 works fine. Is there something I must be missing ? /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --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] sip accounts
Kong wrote: can i know where to start? SIP is such a big topic. Try looking for SIP configuration (sip.conf) in the Wiki, it's got lots of examples. Or you can also try looking it up on google. Flynn ___ --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] [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
On Fri, 2005-10-14 at 15:17 +0200, Patrick de Kok wrote: I would prefer to get it working with i4l at the moment, and migrating later on to CAPI if needed. Thanks for any help you can give me.. And the large number of answers you have received on how to make i4l work doesn't say something? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make error for zaptel
Hi zoltan, I have got the same problem...same error. Seems like the makefile is searching for a modules rule but I looked into Makefile and there is not a 'modules' rule... Have you found a solution? TIA Giorgio Zoltan Szecsei wrote: Bob Goddard wrote: On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote: Hi Bob, Thanks - I'll run with the README idea of yours. Your comment regarding re-boot however is not valid. I also thought that was the case and (as I said on the first line of my message) I specifically rebooted the box. Have to confess I am really flumuxed why the symbolinc link differs from the uname -r name. I cannot see what the problem is with the output of 'uname -r'! I'm saying that I though that if uname -r returns: 2.6.11.4-20a-smp then I would expect that /usr/src/linux would link to linux-2.6.11.4-20a-smp and it does not, it links to linux-2.6.11.4-21.7 see: gl0:/usr/src # ls -la total 499 drwxr-xr-x 17 root root680 2005-06-30 14:46 . drwxr-xr-x 13 root root368 2005-06-30 09:21 .. drwxr-xr-x 3 root root320 2005-05-19 06:53 astcc -rw-r--r-- 1 root root 130943 2005-06-25 19:09 astcc.tar drwxr-xr-x 22 root root 2336 2005-06-23 04:16 asterisk-1.0.8 drwxr-xr-x 7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8 drwxr-xr-x 3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8 drwxr-xr-x 2 root root440 2005-05-23 06:47 btp -rw-r--r-- 1 root root 32975 2005-06-25 19:08 btp.tar drwxr-xr-x 23 root root776 2005-06-12 17:24 dicts drwxr-xr-x 5 root root416 2005-04-01 18:50 gastman -rw-r--r-- 1 root root 332857 2005-06-25 19:08 gastman.tar drwxr-xr-x 25 root root736 2005-06-12 17:29 kernel-modules drwxr-xr-x 2 root root520 2005-06-23 04:11 libpri-1.0.8 lrwxrwxrwx 1 root root 19 2005-06-25 22:01 linux - linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a drwxr-xr-x 3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj drwxr-xr-x 19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj lrwxrwxrwx 1 root root 23 2005-06-25 22:01 linux-obj - linux-2.6.11.4-21.7-obj drwxr-xr-x 7 root root168 2005-06-12 17:43 packages drwxr-xr-x 2 root root 2720 2005-07-01 12:43 zaptel-1.0.8 gl0:/usr/src # uname -r 2.6.11.4-20a-smp So I figured that this may be the reason why the zaptel make is failing. Zoltan If you are saying that you are not running linux-2.6.11.4-20a, then I would say you. Perhaps lilo or grub got corrupted. You should be checking the layout of /boot at least. Bob Goddard wrote: On Friday 01 Jul 2005 13:08, Terry Wade wrote: Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6 Nope, I doubt that. The end user should read /usr/src/linux/README.suse and see how to prepare the kernel for building thirparty modules. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: 01 July 2005 01:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] make error for zaptel Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see listing below). I run make clean ; make linux26 (what about the usual make with no parameters?) and I get a crash. Note that uname -r returns a *different* version of what the linux is linked to (thanks to YOU??) It looks like you updated the kernel but never rebooted. I have tried make clean ; make (no params) and it still crashes. Can anyone offer me some suggestions? - or do I go first to the SuSE list to sort out the uname -r usr/src/linux issue? [...] make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make: *** [linux26] Error 2 gl0:/usr/src/zaptel-1.0.8 # [ Oh for fsck sake, can't people delete old signatures ] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Sending ANI over SIP
I'm running into an issue where subscribers to our service cannot call certain 1-800 numbers if they have a caller id blocked account (restrictcid=yes). This is on Asterisk 1.0.9 and our clients are using Sipura SPA-2002's. Our provider uses a SIP/PSTN gateway, so we hand off SIP to them from Asterisk. The problem appears to be that when an Anonymous call goes out, there is no ANI present in the SIP INVITE. The From: header includes just an IP address -- as does the Contact field. What is the _proper_ way to send ANI via SIP? I am thinking it is Calling-Party-ID but I'm hoping someone can verify this for me as it's not mentioned in the RFC. I've hacked CPID support into Asterisk 1.0.9 but our provider doesn't appear to use it -- ie, if From is Private but CPID is present (with privacy=full), 1-800 numbers still fail because of lack of ANI. My workaround would be to detect numbers which require ANI (1-8XX, 911, etc) and ensure that the From header is always populated for these calls. Mostly I want to find out how things are _supposed_ tow work though. :-) Thanks for any info. Ray -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc. ___ --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] Which asterisk-friendly cards are fax-capable?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Wednesday, October 12, 2005 9:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable? Tom, if you really only have a single-port PRI and can't shell out for a dual, then a T1 fax board is out of the question - it's even more cash. That doesn't leave you with a lot of options except to outsource your faxing. Why not give Lee's stuff a try some weekend when your system's idle? Unfortunately, 1.) I am recently wrestling with stability issues on *, so this takes a back burner... 2.) Our existing analog setup is functioning quite nicely, thank you. 3.) Our system is not busy on the weekends, but it handles our after-hours calls and if it goes down we could have a major problem (we deal with hazardous materials...) Tom ___ --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] 488 Not acceptable here
Perhaps they dont' like the codec you're offering in your INVITE message? Ray On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote: I have been receiving a lot these 488 Not Acceptable Here from a number of providers. What could the problem be? What is the most common cause of this message? ___ --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] '486 Busy here' and 'All Circuits are busynow'
Hi, I've set up IAX FreeWorldDialup on my asterisk server but when I dial my number, I get message '486 Busy Here '. When I dial any other number from my *, it says 'All Circuits are busy now'. What is the problem with my settings? I've followed all the instructions step by step. Hector ___ --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: Re: [Asterisk-Users] IAX or IAX2 ? [SOLVED]
Hi, When I try to load chan_iax2.so, I get the error message The channel name is iax. Yet it provides commands such that begin with iax2 and listens on port 4569. ??? In /usr/lib/asterisk/modules the name of the file ist chan_iax2.so and as far as I understood, I have to enter the file name in modules.conf, right? But if I do this, I get the error chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature Oct 11 10:09:52 WARNING[2288]: loader.c:391 load_modules: Loading module chan_iax2.so failed! Are you sure that the file and the main asterisk binary are from the same source (e.g: debian package)? Yes, it was the same source and the line load = chan_iax2.so was right. But I had to add load = res_crypto.so before. I used grep -r ast_check_signature /usr/src/asterisk-1.0.9/* to find out which other module might use ast_check_signature and so I found res_crypto. Have a nice weekend, Stefan ___ --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] Please Press Any Key to Accept a Call
Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going(for example) to the voice mail on my cell phone. Scenario * Call comes in, outside caller dials "100" * Desk phone for user Joe rings. No answer * Joe's house phone rings. * Joe's wife picks up and hears a voice "Please press any key to accept a call for extension 100." * Joe's wife hangs up. * Joe's cell phone rings. * Joe picks up and hears a voice "Please press any key to accept a call for extension 100." * Joe presses 1 and says "Hello this is Joe". Alternately, in the penultimate step * Cell voice mail picks up. * Voice says "Please press any key to accept a call for extension 100". No keys pressed since it's a voice mail * Call is routed to Asterisk voicemail. It seems straight forward to try multiple locations, but I'm not seeing how to only patch the call through if the user responds with a key press. Thanks, WILL ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Audio from Console but mpg123 from shell works fine.
I get audio from mpg123 at the command line but when I load up asterisk and try to get audio from the console it looks like it's working, and even pauses like it is playing the file but there is no audio coming from the speakers. I have searched and looked through the archives and tried to fix this but I have had no success. This is an onboard Intel card (AC'97) and I also tried an SB Live card with the same result. -Jonathan * Asterisk startup: (asterisk -vvvc) * [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) * Dial 100: * *CLI -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing Playback(OSS/dsp, tones-that-follow-are-for-the-deaf) in new stack -- Playing 'tones-that-follow-are-for-the-deaf' (language 'en') * *** pause while it plays but no audio *** * -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (default, 100, 3) exited non-zero on 'OSS/dsp' Hangup on console * Exit asterisk: (ctrl-c which normally I wouldn't do) * Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (2). * Run mpg123 (Plays fine): (don't need the /dev/dsp, I was just trying to make mpg123 not work to hopefully find out why asterisk doesn't) * [EMAIL PROTECTED] ~]# mpg123 -d /dev/dsp /var/lib/asterisk/mohmp3/TristeAlegriaPromo.mp3 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! Title : 10 - Track 10 Artist: Unknown Album : PROMO Year : Comment: Genre : Club Directory: /var/lib/asterisk/mohmp3/ Playing MPEG stream from TristeAlegriaPromo.mp3 ... Junk at the beginning 49443303 MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo [0:02] Decoding of TristeAlegriaPromo.mp3 finished. [EMAIL PROTECTED] ~]# * Extensions.conf * exten = 100,1,Answer exten = 100,2,Playback(tones-that-follow-are-for-the-deaf) exten = 100,3,Hangup * oss.conf * ; ; Open Sound System Console Driver Configuration File ; [general] ; ; Automatically answer incoming calls on the console? Choose yes if ; for example you want to use this as an intercom. ; autoanswer=yes ; ; Default context (is overridden with @context syntax) ; context=default ; ; Default extension to call ; extension=s ; ; Default language ; ;language=en ; ; Silence supression can be enabled when sound is over a certain threshold. ; The value for the threshold should probably be between 500 and 2000 or so, ; but your mileage may vary. Use the echo test to evaluate the best setting. ;silencesuppression = yes ;silencethreshold = 1000 ; ; On half-duplex cards, the driver attempts to switch back and forth between ; read and write modes. Unfortunately, this fails sometimes on older hardware. ; To prevent the driver from switching (ie. only play files on your speakers), ; then set the playbackonly option to yes. Default is no. Note this option has ; no effect on full-duplex cards. ;playbackonly=yes ___ --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] Don't know what to do if second ROSE component is of type 0x6
FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. We are running a PRI from US LEC (channels 1-10 are B-channels, with channel 24 being the D-channel, and we are only running voice on the PRI.) The PRI is connected directly to our Digium TE110P card, and obviously we are using the zaptel drivers. We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) In our case, it does not seem to affect the stability of our * machine. (However, bear in mind that you may be using parts of * that we do not, and the problem could lie in those parts.) We're handling all PSTN calls via the PRI, except outbound to toll-free which are handed off to an IAX gateway on the Internet. Our employees' desks are connected via the LAN (using Polycom 500/501 SIP phones.) I have a remote extension at home (also SIP) using a Sipura SPA-2000. We did have some stability issues (Asterisk would segfault) when we first moved to CVS. Of course, safe_asterisk handled this and a couple of days later we updated again from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! Tom ___ --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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --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] what should i select ??????????
Anyone know where to get a reasonably priced/chat PoE (powered) switch? For about 5-12 ports? -- Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, October 13, 2005 8:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] what should i select ?? I've done exactly this recently. Frankly with hardphones being as cheap as they are I'd buy them. If you are messing about with analogue adapters etc you'll end up with all sorts of potentional echo problems not to mention the cost of the chanel banks etc. A hardphone will enable you many features that an anologue phone would not. Many hardphones have LCD displays and soft keys which you can assign funtions to such as company directories etc. This is possible in the Analogue world through the use of ADSI but an ADSI compliant phone is often more than a hardphone. There are of course network considerations. Can your network sustain the extra traffic the hardphones will add? Do you have the right physical layer at every station (usually CAT5)? Do you want to do Power Over Ethernet or run a small PSU to the wall for each phone? Only you can answer these questions. For my money, witha new install, I'd go all hardphone. I happen to like the Cisco 7960 which goes for about $260. However there are phones as cheap as $50 in quantity. Check out voipsupply.com for prices etc. Mark ishtiaq Ahmed wrote: hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. - if i use analogue phones in the above case ( we have analogue phones already ) which card should i use.( plzz mention the name of card provided by digium ). i think using some interface cards( for analogue phones and one card for each office) will be cheaper than buying about 200 voip phones. what do u think i will be waiting for ur value able suggestion. i have searched alot for this noone has given me a clear suggestion ( mean to say answer at the max 20% of my question ). -- -- Yahoo! Music Unlimited - Access over 1 million songs. Try it free. http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=36035/*http://music.y ahoo.com/unlimited/ -- -- ___ --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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --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] Please Press Any Key to Accept a Call
I havecoded a new application in Asterisk called app_followme that will do what you're looking for. The caller who made the call originally is also optionally put on hold music while the hunt is going on. There's also planned functionality for blacklisting certain callerIDs so a caller who is blacklisted will think the find-me/follow-me is working, but in reality it's just putting them in a holding pattern and then routing them to voicemail after waiting for about 20-30 seconds evil grin. The code isn't really cleaned up yet from my initial alpha / unit testing on it which is why I haven't put it on the bugtracker yet, but it's quite functional now and I'd like for more people to start testing it if they see a use for this.I'll try to get it up there in the next couple days. It's new functionality, and therefore, won't make it into the 1.2 release of Asterisk, but it doesn't really interfere with much anything else in Asterisk so you should be able to apply the patch cleanly to any recent HEAD branch and probably 1.2 once it's released. BJ On 10/14/05, Will Glass-Husain [EMAIL PROTECTED] wrote: Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going(for example) to the voice mail on my cell phone. Scenario * Call comes in, outside caller dials 100 * Desk phone for user Joe rings. No answer * Joe's house phone rings. * Joe's wife picks up and hears a voice Please press any key to accept a call for extension 100. * Joe's wife hangs up. * Joe's cell phone rings. * Joe picks up and hears a voice Please press any key to accept a call for extension 100. * Joe presses 1 and says Hello this is Joe. Alternately, in the penultimate step * Cell voice mail picks up. * Voice says Please press any key to accept a call for extension 100. No keys pressed since it's a voice mail * Call is routed to Asterisk voicemail. It seems straight forward to try multiple locations, but I'm not seeing how to only patch the call through if the user responds with a key press. Thanks, WILL ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Don't know what to do if second ROSE component is of type 0x6
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6 FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. [snip] We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) Why did you move away from the beta? Tom ___ --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] multi languages
New Language syntax for CVS-HEAD Set(LANGUAGE()=language) How do I use that exactly? Set(LANGUAGE()=en) is for English Set(LANGUAGE()=fr) is for French How do I set it up for Chinese? Set(LANGUAGE()=xx) where do I get the xx ??? The *.gsm files goes for each language into: /var/lib/asterisk/sounds/de and /var/lib/asterisk/sounds/digits/de How do I play the *.gsm files on Windows? bye Ronald Wiplinger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't know what to do if second ROSE componentis of type 0x6
I have been getting that message also. I have been using various versions of CVS head since Feb. 2005. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Don't know what to do if second ROSE componentis of type 0x6 FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. We are running a PRI from US LEC (channels 1-10 are B-channels, with channel 24 being the D-channel, and we are only running voice on the PRI.) The PRI is connected directly to our Digium TE110P card, and obviously we are using the zaptel drivers. We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) In our case, it does not seem to affect the stability of our * machine. (However, bear in mind that you may be using parts of * that we do not, and the problem could lie in those parts.) We're handling all PSTN calls via the PRI, except outbound to toll-free which are handed off to an IAX gateway on the Internet. Our employees' desks are connected via the LAN (using Polycom 500/501 SIP phones.) I have a remote extension at home (also SIP) using a Sipura SPA-2000. We did have some stability issues (Asterisk would segfault) when we first moved to CVS. Of course, safe_asterisk handled this and a couple of days later we updated again from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! Tom ___ --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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --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] 488 Not acceptable here
Quoting Ray Van Dolson [EMAIL PROTECTED]: How can you determine which codecs are acceptable to them? Do they have a way of indicating it? Perhaps they dont' like the codec you're offering in your INVITE message? Ray On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote: I have been receiving a lot these 488 Not Acceptable Here from a number of providers. What could the problem be? What is the most common cause of this message? ___ --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 This message was sent using IMP, the Internet Messaging Program. ___ --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] Don't know what to do if second ROSE component is of type 0x6
I also have the same problem at a customer. It's PRI setup as 5ESS. It doesn;t seem to be hurting anything but it is related to the caller ID. I tried google for docs etc but I came up with nothing I could understand or figure out. Over time I've used both Sangoma and Digium card and it appears with both. I've mostly been running CVS head at this location for several months. The problem still exists in CVS head from around 2 weeks ago. The situation seems related to 800 number inbound only. I have some PRI debug traces where I can call the PRI from my house using the toll free 800 number that is routed to the PRI. I get the second rose 6 error. When I dial the local pilot line that feeds into the PRI the call comes through with no error message. The only difference is the number I use to call the PRI. 800 toll free vs local number. It also seems to affect the Caller ID name. Caller ID number comes throug in both cases. Caller ID Name doesn;t seem to show up when I get a Rose 6 error. I was hoping to put the two traces up next to each other and see what's coming down the D channel differently in the two situations but so far I havn;t had the time... -bill Jeremy Gault wrote: FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. We are running a PRI from US LEC (channels 1-10 are B-channels, with channel 24 being the D-channel, and we are only running voice on the PRI.) The PRI is connected directly to our Digium TE110P card, and obviously we are using the zaptel drivers. We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) In our case, it does not seem to affect the stability of our * machine. (However, bear in mind that you may be using parts of * that we do not, and the problem could lie in those parts.) We're handling all PSTN calls via the PRI, except outbound to toll-free which are handed off to an IAX gateway on the Internet. Our employees' desks are connected via the LAN (using Polycom 500/501 SIP phones.) I have a remote extension at home (also SIP) using a Sipura SPA-2000. We did have some stability issues (Asterisk would segfault) when we first moved to CVS. Of course, safe_asterisk handled this and a couple of days later we updated again from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! ___ --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] ztdummy build problems
Dave Cotton wrote: snip The above is pointing to USB devices are they configured in the kernel and does the machine support uhci? Suggestions? 2.6 ztdummy does not need USB devices. gcc 2.96 don't know if that also will have probs long time since I used it. Upgrade it's free. She's an old dual proc MB and I've had much difficulty doing upgrades else I would. maybe it's time to do the MB upgrade but there is much pain and expense in doing so. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 488 Not acceptable here
Yes. They tell you what's acceptable to them inside the SIP messages you're trading back and forth on the call setup. You can use sip debug within Asterisk to get a closer look at those messages. On 10/14/05, Obelix [EMAIL PROTECTED] wrote: Quoting Ray Van Dolson [EMAIL PROTECTED]:How can you determine which codecs are acceptable to them? Do they have a way of indicating it? Perhaps they dont' like the codec you're offering in your INVITE message? Ray On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote: I have been receiving a lot these488 Not Acceptable Here from a number of providers. What could the problem be? What is the most common cause of this message? ___ --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 This message was sent using IMP, the Internet Messaging Program.___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://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] Enum parse errors
Apparently ENUM now REQUIRES a + at the beginning of the number to query. EnumLookup(+18886532145) No I didn/t see it documented anywhere.. It seems to require it even if there is no + at the beginning of the ENUM record in DNS. We use ENUM to look up extensions and so should not prefix the number with a +. I'm running into errors when using Enum lately. I can't figure out what the problem might be as I've had Enum up and running in the past. I'm running the latest CVS-Head compiled version. I've also tried using the new Enum function with the same results. When doing a lookup on a number that exists in the enum server I get the following results: -- Executing EnumLookup(SIP/MarcSnom-e69d, 18886532145) in new stack Oct 13 22:55:51 WARNING[2208]: app_enumlookup.c:89 enumlookup_exec: The application EnumLookup is deprecated. Please use the ENUMLOOKUP() function instead. Oct 13 22:55:52 WARNING[2208]: enum.c:235 parse_naptr: NAPTR Regex match failed. Oct 13 22:55:52 WARNING[2208]: enum.c:354 enum_callback: Failed to parse naptr :( Oct 13 22:55:52 WARNING[2208]: dns.c:162 dns_parse_answer: Failed to parse result Oct 13 22:55:52 WARNING[2208]: dns.c:208 ast_search_dns: DNS Parse error for 5.4.1.2.3.5.6.8.8.8.1.e164.org -- Executing Dial(SIP/MarcSnom-e69d, ) in new stack Oct 13 22:55:52 WARNING[2208]: app_dial.c:734 dial_exec_full: Dial argument takes format (technology1/number1technology2/number2...|optional timeout) ___ --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] 488 Not acceptable here
Obelix wrote: I have been receiving a lot these 488 Not Acceptable Here from a number of providers. What could the problem be? What is the most common cause of this message? In my experience, that is caused by one side requireing a codec that the other side does not support. ___ --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] Please Press Any Key to Accept a Call
BJ, thanks alot for the coding but I see no reason for it as all you mention is doable currently in Asterisk using some DP magic. Will just search the list and you should find some examples on how to do it (I think there is even an example on the wiki try: http://www.voip-info.org/wiki-asterisk+cmd+dial ) On 10/14/05, BJ Weschke [EMAIL PROTECTED] wrote: I have coded a new application in Asterisk called app_followme that will do what you're looking for. The caller who made the call originally is also optionally put on hold music while the hunt is going on. There's also planned functionality for blacklisting certain callerIDs so a caller who is blacklisted will think the find-me/follow-me is working, but in reality it's just putting them in a holding pattern and then routing them to voicemail after waiting for about 20-30 seconds evil grin. The code isn't really cleaned up yet from my initial alpha / unit testing on it which is why I haven't put it on the bugtracker yet, but it's quite functional now and I'd like for more people to start testing it if they see a use for this. I'll try to get it up there in the next couple days. It's new functionality, and therefore, won't make it into the 1.2 release of Asterisk, but it doesn't really interfere with much anything else in Asterisk so you should be able to apply the patch cleanly to any recent HEAD branch and probably 1.2 once it's released. BJ On 10/14/05, Will Glass-Husain [EMAIL PROTECTED] wrote: Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going (for example) to the voice mail on my cell phone. Scenario * Call comes in, outside caller dials 100 * Desk phone for user Joe rings. No answer * Joe's house phone rings. * Joe's wife picks up and hears a voice Please press any key to accept a call for extension 100. * Joe's wife hangs up. * Joe's cell phone rings. * Joe picks up and hears a voice Please press any key to accept a call for extension 100. * Joe presses 1 and says Hello this is Joe. Alternately, in the penultimate step * Cell voice mail picks up. * Voice says Please press any key to accept a call for extension 100. No keys pressed since it's a voice mail * Call is routed to Asterisk voicemail. It seems straight forward to try multiple locations, but I'm not seeing how to only patch the call through if the user responds with a key press. Thanks, WILL ___ --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] 488 Not acceptable here
SIP DEBUG would show the information in the INVITE. Obelix wrote: Quoting Ray Van Dolson [EMAIL PROTECTED]: How can you determine which codecs are acceptable to them? Do they have a way of indicating it? Perhaps they dont' like the codec you're offering in your INVITE message? ___ --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] ztdummy build problems
On Fri, 2005-10-14 at 08:24 -0700, Bruce Ferrell wrote: She's an old dual proc MB and I've had much difficulty doing upgrades else I would. maybe it's time to do the MB upgrade but there is much pain and expense in doing so. There comes a time... My HP NetServer 5/100 LC looks like it'll have to be laid to rest. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/Cisco Call Manager 3.3
I need to pick all the Asterisk and Cisco People a little. My company has a Cisco Call Manager 3.3, configured via h323 gateways. I have remote users that I want to place a SIP Server on the external WAN and be able to connect their phones to the system and be able to get calls and call people in the office going through the Cisco Call Manager and the h323 router. My only problem is that Cisco Call Manager 3.3 does not support sip trunking. Is there anyway this can be done. Please shed some light on this topic. Thanks. Goran ___ --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] SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?
In Peru you can request Telefonica to provide reversal polarity. Jorge makevuy wrote: Where can I find this information? Faris Raouf wrote: makevuy wrote: Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect Polarity Reversal:yes Detect Disconnect Tone: yes, with the default value. Thanks a lot for your help ;) I've never used one of these (but I'd like one). However, if it is not detecting the disconnect tone, it could be that your telephone service provider is providing a tone is not the same as the one the unit is expecting. For example in the UK you need to change the settings for the disconnect tone from the defaults. Faris. ___ --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] ztdummy build problems
Dave Cotton wrote: On Fri, 2005-10-14 at 08:24 -0700, Bruce Ferrell wrote: She's an old dual proc MB and I've had much difficulty doing upgrades else I would. maybe it's time to do the MB upgrade but there is much pain and expense in doing so. There comes a time... My HP NetServer 5/100 LC looks like it'll have to be laid to rest. More likely that for now I'll just do without ___ --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/Cisco Call Manager 3.3
You can use H.323 on Asterisk and setup CM to use an H.323 gateway to Asterisk, or setup a gatekeeper and have both ends talk to the gatekeeper. I have redundant CM boxes, so I HAD to use a gatekeeper and set it in proxy mode because I had media path issues (the call initiated from one box, but for some reason the media path on outside calls came from the other box and we had one way audio). If there is just one CM box, then you probably don't need a gatekeeper. [EMAIL PROTECTED] wrote: I need to pick all the Asterisk and Cisco People a little. My company has a Cisco Call Manager 3.3, configured via h323 gateways. I have remote users that I want to place a SIP Server on the external WAN and be able to connect their phones to the system and be able to get calls and call people in the office going through the Cisco Call Manager and the h323 router. My only problem is that Cisco Call Manager 3.3 does not support sip trunking. Is there anyway this can be done. Please shed some light on this topic. Thanks. Goran ___ --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 -- Network stuff you didn't know http://www.networkoblivion.com ___ --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] Voicemail - new feature request
Kib Eki wrote: It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Also, one simple thing. Is it possible to listen to my greetings without having to re-record them? ___ --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] Calibrating both RX and TX gain?
I am in the middle of trying to get a milliwatt test line to calibrate the rxgain properly. However, this won't help me with the txgain, will it? How can I properly calibrate the txgain? By ear? Or is there a more scientific method? I contacted Rhino to see if they had any suggestions, and they were able to give me a few. What finally worked was setting the Asterisk gains back to 0 for all channels, then adjusting the gains down on the channel banks themselves for the phone (FXS) interfaces only. A huge improvement! My current adjustements are the following: According to the company that installed the channel bank, there is a 0db and -10db setting on the smart jack for the T1. They claim that this was most likely set to -10db by the ILEC when the T1 was installed, and that would be causing the low audio volume. At the T1 Smartjack point, the level setting is for the T1 digital signal, not the audio level. The two have nothing to do with each other. Does this make sense to anyone? Wouldn't the -10db affect the *digital* levels, not the analog waveform encoded within the digital signal? Correct. I'm still trying to get a milliwatt test line to calibrate from. They claim that they won't give that out to end users because it could fry the T1 card. Sigh. That last statement indicates their level of knowledge; almost zero. A milliwatt generator creates an audio signal at 1,004 hz and 0db. It has nothing at all to do with a T1 signalling, etc. You can yell into a analog telephone set and create audio levels greater then 0 db. Whoever is feeding you the above words apparently has no knowledge of telephony whatsoever. ___ --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] Don't know what to do if second ROSE component is of type 0x6
Tom Rymes ha scritto: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 IIRC, it has to do with rerouted/forwarded calls. I came across that portion of source code when I dealt with call forwarding/deflection. ROSE stands for Remote Operation Service Element; some related information is encoded in information elements as a kind of remote procedure invocation/response, and the support for these things is in libpri is, as far as I know, not complete :) Bye, -- Emanuele ___ --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] Reset telephone IP PHONE 106
Salve Fabio, volevo dirle che può contattare il supporto voismart a [EMAIL PROTECTED] Nel caso non avesse ricevuto risposta, mi avverta che provvederò io stesso a farle avere informazioni. (lo farei ora, ma sono sul treno e non ho accesso ai miei dati) :) greetings, Matteo brancaleoni. On Fri, 2005-10-14 at 11:15 +0200, Fabio Montemaggiore wrote: I have a telephone Voismart IP PHONE 106. I have lost the password of the telephone and therefore I am not able to set up it. How can I do to do a reset of the telephone? -- Come to visit us @ SMAU 2005 From 19 to 23 october Hall 12 Booth H22 Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] warning message when reloading chan_sap.so
Does anyone know what this error means? WARNING[12632]: chan_zap.c:10084 setup_zap: Ignoring signalling Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 15, Issue 85
Does anyone know why you must sometimes dial a number twice to get it to connect? We have IP phones and a TDM400P Digium card that connect to 2 PSTN lines. Some numbers work fine and then other don't. What gives? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does anyone Know if tha avaya 4621 IP phone work wiht asteisk?
Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? if it work it has featuras working Thanks Ignacio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Please Press Any Key to Accept a Call
BJ, Great news - glad to hear it. I think the key thing I'm looking for is that this is all transparent to the caller. I want them to hear nice hold music while the user is searched for, and only be directed to the physical extension if the correct person picks up. (e.g. no third party voice mail). CF - That sounds good, I'll search the archives. Possibly an application would still be of service if it simplifies the dial plan. Look forward to trying both approaches. Best, WILL ___ --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] IAXy Port number. Repost
The file used with a Digium IAXy device: iax.conf has the line: port=5036 (I also use the bindaddr=192.168.1.91 entry) but when Asterisk talks to the IAXy device it used port 4569 (from tcpdump). How are the port numbers assigned? What tells the IAXy which port to use. The IAXy provisioning file iaxy.conf file does not specify a port (I'm configured for static IP). I'm trying to use the IAXy device across a VPN. ___ --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] Calibrating both RX and TX gain?
On Fri, Oct 14, 2005 at 11:38:00AM -0600, Rich Adamson wrote: A milliwatt generator creates an audio signal at 1,004 hz and 0db. It has nothing at all to do with a T1 signalling, etc. You can yell into a analog telephone set and create audio levels greater then 0 db. Whoever is feeding you the above words apparently has no knowledge of telephony whatsoever. I got a similar run-around from my telco (SBC). The customer and technical service people had no idea that this type of line existed. They sent me to the tech dispatch people, who seemed to know what kind of line this was, but said, we don't have these kind of numbers... that we give out to anybody. It's just a number that plays a tone at a reference volume when you call it. Maybe you can get the number from some area telco consultants/installers. -- Shaw Terwilliger [EMAIL PROTECTED] SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 15, Issue 85
Does anyone know why you must sometimes dial a number twice to get it to
connect? We have IP phones and a TDM400P Digium card that connect to 2 PSTN
lines. Some numbers work fine and then other don't.
What gives?
The most frequent cause that I've heard of is that asterisk starts pumping
out the dialed digits too soon, and the central office equipment only
catches the last seven of eight digits, etc.
Try inserting a w in the Dial statement something like this:
exten = _21X,1,Dial(Zap/g1/w${EXTEN})
You won't find that documented very well in 'show application dial', but
the w injects something like a 250 millisecond delay before dialing.
You can use multiples if needed (www) for a longer delay.
___
--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] DID on analog line
Paul wrote: Last time I checked analog DID trunks were expensive both NRC and MRC. Depends on your ILEC/CLEC. Here is Vancouver they are the same price as non-DID trunks with DID numbers $2/ea in quantities 1000 (from at least one CLEC). I have heard of CLECs in the US where DIDs are a tenth of this cost. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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] USB phone for Linux?
Michael Van Donselaar wrote: On Thu, 13 Oct 2005 08:41:17 -0400, Paul [EMAIL PROTECTED] wrote: Tony Mountifield wrote: Hi, Can anyone recommend a USB phone that can be used under Linux, either interfacing directly with Asterisk in some way, or using a soft phone program on Linux that doesn't need screen interaction (only using the phone's keypad)? The idea is to be able to plug it into the USB port of an Asterisk box in a rack, where screen, kbd and mouse may not be available. Thanks in advance! Tony Find me a USB phone with sufficient hardware docs available and I will see what I can do. I could use the same type of thing. I have remote customer servers and would love to have them setup so my contractor tech can just plug in and become extension on my pbx here. Tigerjet makes a USB handset that is based on the same chipset as the S100U. In fact, it looks like there's enough info in the wcfxsusb.[ch] files in zaptel to get the keypad running. I like the AU100 USB phone a lot better, but it looks like it will be windows only. (ironically the AU100 was a lot easier to get working with iaxComm than the Tigerjet phone) What I would do is base the softphone on something like iaxclient. I would have it launched when the usb hotplug was seen. I suppose this could be initially done with 2 devices. One would be a good usb headset and the other would be a keypad with lcd display. I don't see any lcd display on the $34 phone. Maybe the $34 usb phone adaptor is a better starting point for me because it says caller ID is supported. I can plug a $12 phone that supports caller ID with name into the thing. ___ --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] How to rewrite a CALLERID on outgoing calls
Hello all, is here anybody who have any idea how i can insert a script or program to rewrite a callerid with special rules. This ist necessary because of many moving mobile offices who changes the telefonenumbers in short time distances. I've found the SetCIDName feature. But this doesn't work in my relation. The phones from wich the calls are, are connected via OH323 Module and registered on a gnugk Gatekeeper. Thanks Hans-Peter Straub -- ---* I-NetPartner GmbH Hans-Peter Straub Seewiesenstrasse 12 D-73054 Eislingen -- Phone: +49 7161 9849955 Fax: +49 7161 9849956 -- eMail: [EMAIL PROTECTED] Web: http://www.I-NetPartner.de ---* ** Informieren Sie Sich über ** -- GigaLan -- ** das Funknetz im Filstal ** http://www.GigaLan.de ---* -- PGP-ID: 24557EED PGP-Key: http://www.i-netpartner.de/hps.asc PGP-Fingerprint: 51F2 31E4 4361 1B7F 8648 60D9 FC1A 68D2 2455 7EED -- ___ --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 rewrite a CALLERID on outgoing calls
Am Freitag 14 Oktober 2005 21:22, Hans-Peter Straub schrieb: Hello all, is here anybody who have any idea how i can insert a script or program to rewrite a callerid with special rules. This ist necessary because of many moving mobile offices who changes the telefonenumbers in short time distances. I've found the SetCIDName feature. But this doesn't work in my relation. The phones from wich the calls are, are connected via OH323 Module and registered on a gnugk Gatekeeper. Hello again, sorry i made a fault in my tests. SetCIDName does now work for me fine. Thanks Hans-Peter Straub -- ---* I-NetPartner GmbH Hans-Peter Straub Seewiesenstrasse 12 D-73054 Eislingen -- Phone: +49 7161 9849955 Fax: +49 7161 9849956 -- eMail: [EMAIL PROTECTED] Web: http://www.I-NetPartner.de ---* ** Informieren Sie Sich über ** -- GigaLan -- ** das Funknetz im Filstal ** http://www.GigaLan.de ---* -- PGP-ID: 24557EED PGP-Key: http://www.i-netpartner.de/hps.asc PGP-Fingerprint: 51F2 31E4 4361 1B7F 8648 60D9 FC1A 68D2 2455 7EED -- ___ --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] soxmix generating mute files
Hello All, I am trying to use soxmix to merge two wav files generated by monitoring calls from a queue, since it generated two files (in out). When I run soxmix file1.wav file2.wav mixedfile.wav, although file1.wav and file2.wav are good, mixedfile.wav is file with the same size as file2.wav, but totally mute. Any clues? Thank you!Dov ___ --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] Busy not jumping n + 101 anymore
I recently upgraded my Asterisk system to the latest CVS-HEAD Asterisk CVS HEAD built by[EMAIL PROTECTED] on a i686 running Linux on 2005-10-12 13:34:09 UTC Ever since this upgrade, the system is jumping n+101 if it gets a busy on a Dial command, it is now proceeding to the next priority (n+1) Has something changed with this? Is there a way to change it back? Thanks, Bill ___ --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] Busy not jumping n + 101 anymore
On 10/14/2005, William M. Sandiford [EMAIL PROTECTED] wrote: I recently upgraded my Asterisk system to the latest CVS-HEAD Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-10-12 13:34:09 UTC Ever since this upgrade, the system is jumping n+101 if it gets a busy on a Dial command, it is now proceeding to the next priority (n+1) Has something changed with this? Is there a way to change it back? So glad to see you read the documentation... Try scanning UPGRADE.txt A lot has changed. Brett ___ --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] sip accounts
Hi, Try add [1234] ... host=dynamic or host=xxx.xxx.xxx.xxx (the client's IP) ... ... AK On 10/14/05, Kong [EMAIL PROTECTED] wrote: hi, i facing a problem here. in my sip.conf, i specify a account like this,[1234]type=friendcontext=from-sip username=1234secret=1234nat=nocanreinvite=yesdtmfmode=info[EMAIL PROTECTED]disallow=allallow=ulawso i am able to login with username 1234 and password 1234but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus anybody can just use mysystem to call others.. lets say i do have set that some certainaccount can make some certain calls only.how can i solve this problem? thank you.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID on analog line
I got my next cabinet convinced to use his digital PRI to do the test, which would be what we would really use after all. I have an idle digum T1 card sitting, so lets see if I can get that going. Thanks for all your help. I learn something new everyday. -apu On 10/14/05, George Pajari [EMAIL PROTECTED] wrote: Paul wrote: Last time I checked analog DID trunks were expensive both NRC and MRC.Depends on your ILEC/CLEC. Here is Vancouver they are the same price as non-DID trunks with DID numbers $2/ea in quantities 1000 (from atleast one CLEC). I have heard of CLECs in the US where DIDs are a tenthof this cost.--George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists1 877 NET VOIP (638 8647 x102) www.netvoice.cawww.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Outbound registration expirey
Hi list! I´m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can´t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742 handle_response_register: Outbound Registration: Expiry for gvt.com.br is 60 sec (Scheduling reregistration in 45 s) I´m using asterisk-1.2.0-beta and the sip.conf parameters about registration: defaultexpirey=1200 registertimeout=1200 There is any way to make asterisk follow the 1200 seconds I´m trying to tell? Could be something happening out of my unit but at the provider network? Thanks in advance, Ricardo Poppi. ___ --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] Busy not jumping n + 101 anymore
[EMAIL PROTECTED] wrote:
On 10/14/2005, William M. Sandiford [EMAIL PROTECTED]
wrote:
Ever since this upgrade, the system is jumping n+101 if it gets a busy
on a Dial command, it is now proceeding to the next priority (n+1)
Has something changed with this? Is there a way to change it back?
So glad to see you read the documentation...
Try scanning UPGRADE.txt
A lot has changed.
We've had the same problem here ever since we upgraded to CVS-HEAD.
When someone placed a call to a number that was busy, they would just
receive the call cannot be completed recording we have setup at n+1.
Not to sound nitpicky or hateful, but I just reviewed UPGRADE.txt again
here and I don't see anything about it. If it is in there, could you
please point it out to me? (Seriously, as I didn't see it.) If it
isn't, someone with CVS access should probably add it in.
Now, I will say that I'm assuming (from the new behavior and the show
application dial output) that one should now be using the ${DIALSTATUS}
variable to handle these conditions. (i.e. from your dial, make n+1 be a
Goto(s-${DIALSTATUS}) command, and create s-BUSY, s-CONGESTION, etc. in
the same context.) Once I get around to updating our dialplans, that's
what I plan on doing.
Someone please correct me if I am wrong. *dons asbestos armor, just in
case*
Jeremy
--
Jeremy Gault, KD4NED[EMAIL PROTECTED]
Network Administrator, WinWorld Corporation
Member: Bradley County ACS/RACES/SkyWarn
voice: +1.423.473.8084 fax: +1.423.472.9465
Want a free GMail invite? E-Mail me and let me know!
___
--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] DID on analog line
George Pajari wrote: Depends on your ILEC/CLEC. Here is Vancouver they are the same price as non-DID trunks with DID numbers $2/ea in quantities 1000 (from at least one CLEC). I have heard of CLECs in the US where DIDs are a tenth of this cost. Yep. We're using US LEC here (with a PRI) and they charge us ~$4/month for a block of 20 DIDs. I'm not sure if they would do any of the analog DID stuff, though. Actually, I don't think you can purchase straight analog lines from them (unless you co-lo at their switch.) Instead, if you need analog, they'll bring in a T1 and setup a channel bank for you. I've always wondered why some places charge so much for DIDs, though. Jeremy -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --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] 2 POTS to
Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use asthe 8FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog switch for 2 incoming POTS to 8 POTS is a better solution, i would appreciate ifyou could point me to posibles solutions. But I would prefer not to lose the IP option, so later i could ad some ip phones, or softphones, and be able to make calls to FWD numbers, etc, through my internet connection. Regards, tia Claudio ___ --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] Busy not jumping n + 101 anymore
Ever since this upgrade, the system is jumping n+101 if it gets a busy
on a Dial command, it is now proceeding to the next priority (n+1)
Has something changed with this? Is there a way to change it back?
So glad to see you read the documentation...
Try scanning UPGRADE.txt
A lot has changed.
We've had the same problem here ever since we upgraded to CVS-HEAD.
When someone placed a call to a number that was busy, they would just
receive the call cannot be completed recording we have setup at n+1.
Not to sound nitpicky or hateful, but I just reviewed UPGRADE.txt again
here and I don't see anything about it. If it is in there, could you
please point it out to me? (Seriously, as I didn't see it.) If it
isn't, someone with CVS access should probably add it in.
Now, I will say that I'm assuming (from the new behavior and the show
application dial output) that one should now be using the ${DIALSTATUS}
variable to handle these conditions. (i.e. from your dial, make n+1 be a
Goto(s-${DIALSTATUS}) command, and create s-BUSY, s-CONGESTION, etc. in
the same context.) Once I get around to updating our dialplans, that's
what I plan on doing.
Someone please correct me if I am wrong. *dons asbestos armor, just in
case*
I remember seeing the cvs-head code change (think yesterday), and if
memory serves correctly, there were comments added to the extensions.conf
file relative to the n+101 behavior. Take a look in there.
I don't recall seeing any upgrade.txt changes, and I haven't completed
a cvs update for about a week so really can't check.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 POTS to
I dont think the Quintum hardware supports SIP devices (just SIP trunks). -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudio Canseco Sent: Friday, October 14, 2005 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 2 POTS to Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use asthe 8FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog switch for 2 incoming POTS to 8 POTS is a better solution, i would appreciate ifyou could point me to posibles solutions. But I would prefer not to lose the IP option, so later i could ad some ip phones, or softphones, and be able to make calls to FWD numbers, etc, through my internet connection. Regards, tia Claudio ___ --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
