Re: [Asterisk-Users] Receptionist Phones
2006/3/29, John Novack [EMAIL PROTECTED]: The reality is, of course, that telephone systems have provided thisfunction for many years. A DSS/BLF is available on MANY so called legacysystems, so until this function is readily available , customers that require a receptionist will continue to go elsewhere.Perhaps it is time to rethink the way data is exchanged between the CPUand the DSS/BLF?As someone said a very long time ago:Results, not excuses. With user count growing, I think receptionist could evolve from hardware to hardware-software combination the same as receptionist job changes from assisting call transfers (check if someone is available before transfer) to blind call transfering (forward anyway and take the call back if nobody answers). If my understanding is correct, in the later case, a receptionist doesn't really need BLF : he or she simply forward the call.He or she mainly needs a directory application helping him or her to find the right person within the organisation. And I don't think anyone could have the patience to harphone BLF labels every 2 weeks to keep up large site permanent user moves, adds and changes. So the perfect receptionist application hardware-software combination should include a mix between directory application and softphone, and provide comfortable hardware to support these.My opinion is I don't think market trends are at works now to make this perfect combination happen anytime soon : - from my point of view, it could take years to gather inputs from receptionist around the world to provide them an effective software-hardware combination.- no one around the world really targets receptionist tools market (is it a market ?) : some companies sell headphones or hardphones but receptionnist account for such a tiny part of sales that these companies cannot really hear receptionists demands and design specific products. - even if someone ever decide to focus on this, it would be difficult for someone to convince companies to improve receptionist tools once receptionist are trained and used.Maybe, a standard PC+headphone + a couple of software would be the best way to go ? Even on that, obstacles remain such as :- how do you monitor legacy PBX users along Asterisk users ?- how do you monitor a distant Asterisk server whitout any Data link between both locations ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to check if a phone / line is used?
In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? Sip show peers shows me just if it is on-line, but not if it is in a call or not. In the dialplan I could dial the number and if it is busy, it would go to the Voicemail for unavailable or busy. I expect that there is just a test function as well, without trying to call. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff does not work with TDM400P
Title: bristuff does not work with TDM400P Hi- we are having issues with quadBRI card which does not work together with TDM400P. We've tried to hunt the problem and here is the scenario: 1) starting asterisk with tdm400P and two FXS modules (two phones) 2) pickup first phone and dial the second one works great 3) hangup 4) pickup second phone and tried to dial the first phone - no luck - asterisk does not recognize DTMF of the dialing numbers and does not initiate call 5) restarting asterisk 6) go back to 4 and works! 7) go back to 2 and does not work again - same asterisk does not recognize DTMF of the dialing numbers Vanilla asterisk works just fine. The above scenario works even quadBRI card is removed it must be problem of bristuff patches. Do you have any hints what can be wrong? We've tried latest bristuff-0.3.X series. Thanks. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode
I had conversation with Welltech support and I got this description (I can't send attachment through the list): The 380x has a routing table function. There are two default route exist in the routing table, one is for IP incoming call another is for FXO incoming call, the IP call will be routed to FXO and the Call from FXO side will be routed to IP side. If the 380x got an incoming call from IP side and the line number is not it's local SIP number. Then the 380x will forward this number to the FXO which is based on the default route, and the FXO will dial this number through PSTN automatically. User just need to dial the destination PSTN number, this is called one-stage dialing. For more information about one-stage dialing, please refer to the attachment. If the 380x got an incoming call from IP side, and the line number is the FXO's local SIP number, then the FXO will answer this call and user will hear dial tone from PSTN side, then they should re-dial the destination PSTN number. User should dial number twice, so this is called two-stage dialing. artifex On 3/31/06, Erick Perez [EMAIL PROTECTED] wrote: one-stage calling function? On 3/30/06, kevin ling [EMAIL PROTECTED] wrote: Yes, Same configuration as Martin. 1.for incoming call just set the 3804 hotline to one sip extension number. 2.for outgoing call, you just using regular dail command to pass the phone number to 3804 (3804 is a 4FXO port device, the call from ip side always pass to FXO Port). You can telnet to the 3804 and enable the one-stage calling function. Regards, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How is Teliax ?
Lonnie Abelbeck wrote: asterisk at anime.net writes: On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote: I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers. I have had issues with termination on teliax. Callers tell me I sound choppy to them. Teliax origination has no problems at all strangely enough. If you used SIP instead of IAX2 with Teliax you will have better quality calls. The 'choppy' sound occurs with IAX2 and not SIP at Teliax. I can recommend Teliax, but use SIP. But I _am_ using SIP. I tried all the various teliax gateways including the beta test ones and had choppiness with all of them. As I said before, teliax origination had no choppiness problems at all. Only termination had issues. I had no problems - termination or origination - with junction networks, despite the fact they had 3x higher latency than teliax. JN is more expensive than teliax though. Also, I have talked to others who had similar choppiness problems with teliax. So it's not just me. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No voice heard in festivalassociated with asterisk!!!
Hi all I complile asterisk 1.2.4 successfully.I install festival successfully and i configure asterisk to work with festival.But When i call the festival extension configured in extensions, the festival application is executed well (i see it in the log) and must read the text (hello world).But i'm hear no voice. What's the problem? Thanks Serge ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing SIP Failover
I am trying to write a outgoing Macro which has some sort of failover for failing SIP connections. For example... Try Outgoing SIP Provider 1 - No Route to Destination Try Outgoing SIP Provider 2 - Congested Try Outgoing SIP Provider 3 - Success and connect.. Everything I try doesnt work. Even if you can just point me to a good website where I can get this information.. Kind Regards, Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum Tenor DX4060
Has anyone linked the Asterisk to the Quintum Tenor DX4060? If yes I would appreciate any valuable information to do this in anyway. Cheers Stephen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Hylafax, on the same box
On 08:54, Fri 31 Mar 06, Olivier Krief wrote: Why don't everybody use chan-capi ? All our E1 interface use the zaptel driver, so impossible to use chan_capi for them. We use Sangoma cards, and the wanpipe driver for those cards is a zaptel interface for asterisk, not a capi one. There is your why?. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Benchmarking an Asterisk Server with 14k users
On 03/31/06 00:28 Stefan-Michael. Guenther (in-put GbR) said the following: To make it clear: We don't want to compare the three system against each other. The asterisk server is running on a completely different hardware. We what are the hardware and OS specs for the asterisk server ? this will form the crux of what you're testing. 7,000 simultaneous calls seems high for a single server to handle, you may need to build a cluster of asterisk servers to handle this. signate has claimed 5,000 simultaneous calls on their asterisk based product. -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone - Tenovis PBX - QSIG - Asterisk - SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of type 0x6 Don't know what to do if second ROSE component is of type 0x6 Don't know what to do if second ROSE component is of type 0x6 Don't know what to do if second ROSE component is of type 0x6 !! Unknown IE 49 (cs5, Unknown Information Element) !! Unknown IE 50 (cs5, Unknown Information Element) -- Accepting call from '1311' to '03' on channel 0/1, span 1 -- Executing Goto(Zap/1-1, default|8403|1) in new stack -- Goto (default,8403,1) -- Executing NoOp(Zap/1-1, 8403) in new stack -- Executing Dial(Zap/1-1, SIP/8403) in new stack -- Called 8403 -- SIP/8403-af88 is ringing -- SIP/8403-af88 is ringing -- SIP/8403-af88 is ringing -- SIP/8403-af88 answered Zap/1-1 == Spawn extension (default, 8403, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --- However, the opposite way (i.e. SIP phone - Asterisk - QSIG - Tenovis PBX - traditional phone) doesn't work at all. I get the following messages: --- -- Executing Dial(SIP/8403-5b0f, Zap/g1/1311) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/1311 Mar 31 12:45:34 WARNING[23193]: chan_zap.c:7792 pri_fixup_principle: Call specified, but not found? Mar 31 12:45:34 WARNING[23193]: chan_zap.c:9046 pri_dchannel: Unable to move channel 1! Don't know what to do if second ROSE component is of type 0x6 XXX Invalid Progress indicator value received: 14 -- Zap/1-1 is ringing Don't know what to do if second ROSE component is of type 0x6 XXX Invalid Progress indicator value received: 14 -- Zap/1-1 answered SIP/8403-5b0f -- Hungup 'Zap/1-1' == Spawn extension (default, 1311, 1) exited non-zero on 'SIP/8403-5b0f' --- The called phone does NOT ring and I get some kind of busy tone on the SIP phone. As Asterisk says answered and the SIP phone counts the ellapsed time, it seems like the call has succeeded from the SIP phone's and Asterisk's perspective, i.e. the Tenovis PBX generates the busy tone?! The Tenovis service guy told me that I need to tell him the correct QSIG settings for the PBX: --- QBC QSIG B-Kanal zyklisch *QBS QSIG Leistungsmerkmale sperr. QBADI QSIG barr. suppl. serv. addit. indication QBANI QSIG barr. suppl. serv. ani QBCCC Rueckruf komplett sperren QBCFA senden (De)Aktivier. sperren *QBCFC RUL Pruefung sperren QBCFF senden RUL-Facility sperren QBCFL RWL spaetes Ausloesen sperren QBCHN QSIG Gebuehren Anforderung Netz *QBCHR Anfordern Gebuehren sperren QBCII Call Intrusion Invoke sperren QBCLI QSIG barr. suppl. serv. call linkage QBCMN QSIG barr. suppl. serv. CoMmon info extension QBCMS QSIG barr. suppl. serv. CoMmon info solic.serv. QBCMU QSIG barr. suppl. serv. CoMmon info unsolic.serv. QBCNF QSIG barr. suppl. serv. conference *QBCOI Anklopfen sperren QBCPI QSIG barr. suppl. serv. call park QBCPR QSIG barr. suppl. serv. call park retrieve QBCST QSIG barr. suppl. serv. csta *QBCTF senden Umlege-Facility sperr. QBCTM TLC line code QBDAS Sperren der Distinctive-Alerting Signalisierung QBDCH QSIG barr. suppl. d channel supervision QBDMI QSIG barr. suppl. serv. DSS module invoke QBDNW QSIG barr. suppl. serv. csta QBDSP QSIG barr. suppl. serv. display QBMMI SS minimail invoke barring QBMWI QSIG barr. suppl. serv. messg. wait. invoke QBNIA Namensanz. geruf. Tln sperren QBNIB Namensanz. bes. Tln sperren QBNIC Namensanz. verbu. Tln sperren QBNIO Namensanz. ruf. Tln sperren QBNMW NWR Message Waiting im Netzwerk sperren QBNWP QSIG netzweite Partner sperren QBPDI QSIG barr. suppl. serv. post dial info QBPRI Ersatzwege-Suche sperren *QBPUP QSIG barr. suppl. serv. pick-up QBRCI QSIG barr. suppl. serv. recall invoke QBRPE QSIG barr. suppl. serv. radio paging equip. QBSEA QSIG Dienstkennung erweiterte Adressierung QBSME QSIG Dienstkennung Herstel. Erweiterung QBSOM QSIG
[Asterisk-Users] How do you perform a Variable Substitution In Asterisk
Dear Group;
I have a requirement to pass the ${SIPDOMAIN} variable from Server A to
Server B over IAX2. Basically Server A is an Internal (*) and Server B
is an External (*) in the DMZ.
On Server A I do the following;
[SIPOUT]
exten = _6.,1,SetVar(DS=${EXTEN}%${SIPDOMAIN})
exten = _6.,2,Dial(IAX2/bxx:[EMAIL PROTECTED]/${DS})
On the CLI I get;
-- Executing Dial(SIP/phone6-bd3d,
IAX2//bxx:[EMAIL PROTECTED] /6shad%xxx..com) in new stack
This comes through over IAX2 and I can strip the 6 and send the call out
via SIP to my SIP proxy.
The only item missing is to substitute the % with @. Can this be done
natively in Asterisk? My production version is Asterisk CVS-v1-0-07.
I have read through
http://www.voip-info.org/wiki/view/Asterisk+variables and could see no
obvious method for this.
Many Thanks
Shad Mortazavi
-
Nexus Group Technical Manager
n|m Nexus Management Inc
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RE: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC,and bristuff - a general questionto clear up my understanding. I have 2 different instalations with 1 Billion HFC Card (1port), and 1 TDM400. Asterisk 1.0.10+bristuff+florz patch. Only issue is that you must load all modules (wcfxs, zaphfc) before runing ztcfg, otherwise nothing works. Everything works ok, even faxing. Julian. On 3/30/06, Chris Earle [EMAIL PROTECTED] wrote: What? After hours of searching for anything to help me, I found this comment about zaptel cards in systems with bristuff-cards (junghanns for me in this case) I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine? thanks :-S Zoa [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] snip We stopped with the bristuff as bristuff will break any other zaptel cards in the same system. (pri seems logical, why the tdm card also broke is unknown to me). snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing SIP Failover
I know that astcc has that feature built in. In it you specify your diffrent routes and the order. Have a look at it. --- Steve Ducat [EMAIL PROTECTED] wrote: I am trying to write a outgoing Macro which has some sort of failover for failing SIP connections. For example... Try Outgoing SIP Provider 1 - No Route to Destination Try Outgoing SIP Provider 2 - Congested Try Outgoing SIP Provider 3 - Success and connect.. Everything I try doesnt work. Even if you can just point me to a good website where I can get this information.. Kind Regards, Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please Help Test Quad PRI Using NFAS
I liked the music --- Andrew Latham [EMAIL PROTECTED] wrote: hint... - listen to the queue for a bit On 3/30/06, Melcon Moraes [EMAIL PROTECTED] wrote: Are you gonna answer me? I'm the first in line and no answer! :) []'s MM -Original Message- From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Thu, 30 Mar 2006 17:13:46 -0400 Delivered: Thu, 30 Mar 2006 19:03:15 Subject:[Asterisk-Users] Please Help Test Quad PRI Using NFAS Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1143756585.491294.13822.aldavila.hst.terra.com.br,4136,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
Asterisk 1.2.4 + bristuff-0.3.0-PRE-1l + libpri-1.2.2 + zapte-1.2.3 + *CLI zap show status Description Alarms IRQ bpviol CRC4 quadBRI PCI ISDN Card 1 Span 1 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 2 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 3 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 4 [TE] (caˇ OK 0 0 0 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 and it seems to work. Only sip phones connected to *PBX (by gateways or ethernet) as i remember installation process: 1. bristuff patching asterisk/libpri/zaptel 2. libpri/zaptel/asterisk install 3. zaptel/quozap/wctdm modules installation Runs on Debian 3.1. kernel 2.6.15.4 but i have problem: when SIP hardphone hangup connection (SIP/ - Zap/) asterisk don't send Q.931 DISCONNECT message, and i don't have any idea how to fight with that. Filip D. Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC,and bristuff - a general questionto clear up my understanding. I have 2 different instalations with 1 Billion HFC Card (1port), and 1 TDM400. Asterisk 1.0.10+bristuff+florz patch. Only issue is that you must load all modules (wcfxs, zaphfc) before runing ztcfg, otherwise nothing works. Everything works ok, even faxing. Julian. On 3/30/06, Chris Earle [EMAIL PROTECTED] wrote: What? After hours of searching for anything to help me, I found this comment about zaptel cards in systems with bristuff-cards (junghanns for me in this case) I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine? thanks :-S "Zoa" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... snip We stopped with the bristuff as bristuff will break any other zaptel cards in the same system. (pri seems logical, why the tdm card also broke is unknown to me). snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Filip Drągowski Mobile: +48(0)500 054045 E-mail: [EMAIL PROTECTED] ___ ONTP.NET Tomasz Karczewski Aleja Wojska Polskiego 33 pokoj 122 65-077 Zielona Góra, Poland Mobile: +48(0)501 653395 Office: +48(0)68 4141018 Fax:+48(0)68 4141017 http://www.ontp.net ___ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Hi Johann, Johann Hanne [EMAIL PROTECTED] writes: Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. I tried with asterisk 1.2.2 against Alcactel 4400 on Monday. We had partially success. But at a specific config on the Alcatel side, the called number was not set by the SETUP message but via INFORMATION messages. Well, libpri doesn't like it this way. AFAIR, libpri does Q.SIG basic call, so you should set the Tenovis also to basic call. If this doesn't help, please run a pri debug span 1 while you make calls and post the output. My conclusion with Q.SIG: do not use it at this implementation level. YMMV. cu, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Howto Cut the first digit
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
but it didn't work, my problem is the delemiter, I have no delemiter,
the default is - but how to use the function cut() without an delemiter?
Just snip the first digit of a phonenumber.
MfG,
Christian Reelfs
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[Asterisk-Users] Howto cut the first digit
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
but it didn't work, my problem is the delemiter, I have no delemiter,
the default is - but how to use the function cut() without an delemiter?
Just snip the first digit of a phonenumber.
MfG,
Christian Reelfs
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RE: [Asterisk-Users] Realtime Users/Peers/Friends - Ick
Doug this is in no way an offense to you but I think we need to start the asterisk booze fund. This will be for all of us that have ups and downs in working on getting asterisk set up. I for one have my friend Johny Walker right by my side when ever it gets to me. --- Douglas Garstang [EMAIL PROTECTED] wrote: I just tried setting all my phone's to type=peer. Seems to break everything. I heard that type=friend was going to be phased out in upcoming releases of Asterisk. I sure hope the developers have thought this though. Reason? Well our phones send calls to Asterisk through OpenSER. We have multiple OpenSER systems that the calls may hit Asterisk from. When I set all my phone accounts in my sip peers table to type=peer, Asterisk no longer matches against them, and instead matches against the OUTGOING OpenSER proxy entries (we place calls to the PSTN through OpenSER too). I guess it does this because it matches the source IP address of the INVITE against the host= value against the proxy in sip.conf. Asterisk responds with: --- (14 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to xxx.187.142.205 : 5060 (non-NAT) Found no matching peer or user for 'xxx.187.142.205:5060' Bottom line is that incoming calls from phones through OpenSER no longer match against the individual phones accounts, but against the outgoing (they have PEER!) OpenSER proxy entries. Now that I think about it, why the heck do incoming calls match against a type=peer entry anyway? I thought a peer was for outgoing calls only??? Is this the way it's going to work in some future release of Asterisk? Btw, I tried setting the phones to peer because I don't know what the frig I'm doing. Doug -Original Message- From: Douglas Garstang Sent: Wed 3/29/2006 10:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] Realtime Users/Peers/Friends - Ick I've been going in circles for a few weeks now with Realtime SIP. My extconfig.conf has: sipusers = mysql,dbname,ast_sip_users sippeers = mysql,dbname,ast_sip_users When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as 'RFC3581', eventhough I have nat set to NO for every friend in the ast_sip_users table. In short, phones make and receive calls, so they should be defined as type=friend, right? Should I point sipusers and sippeers from extconfig to the same table? Why does 'extconfig' have sipusers and sippeers? This is driving me nuts! Is this actually documented anywhere? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto cut the first digit
Christian Reelfs [EMAIL PROTECTED] writes:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
Look at README.variables! It says:
,
| The format for removing characters from a variable can be expressed as:
|
| ${variable_name[:offset[:length]]}
|
| If you want to remove the first N characters from the string assigned
| to a variable, simply append a colon and the number of characters to
| remove from the beginning of the string to the variable name.
|
| ;Remove the first character of extension, save in number variable
| exten = _9X.,1,Set(number=${EXTEN:1})
`
cu,
Wolfgang
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Re: [Asterisk-Users] Howto cut the first digit
http://www.voip-info.org/wiki-Asterisk+variables
section: substrings
F.
Christian Reelfs [EMAIL PROTECTED] writes:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
Look at README.variables! It says:
,
| The format for removing characters from a variable can be expressed as:
|
| ${variable_name[:offset[:length]]}
|
| If you want to remove the first N characters from the string assigned
| to a variable, simply append a colon and the number of characters to
| remove from the beginning of the string to the variable name.
|
| ;Remove the first character of extension, save in "number" variable
| exten = _9X.,1,Set(number=${EXTEN:1})
`
cu,
Wolfgang
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--
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Mobile: +48(0)500 054045
E-mail: [EMAIL PROTECTED]
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Aleja Wojska Polskiego 33 pokoj 122
65-077 Zielona Góra, Poland
Mobile: +48(0)501 653395
Office: +48(0)68 4141018
Fax:+48(0)68 4141017
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Re: [Asterisk-Users] Howto cut the first digit
On Fri, 2006-03-31 at 14:03 +0200, Christian Reelfs wrote:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
Try just:
exten = _[0-9].,1,Dial(Zap/g1/${EXTEN:1})
Pete
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RE: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again but only one direction. -David From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Filip Drągowski Sent: Friday, March 31, 2006 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding. Asterisk 1.2.4 + bristuff-0.3.0-PRE-1l + libpri-1.2.2 + zapte-1.2.3 + *CLI zap show status Description Alarms IRQ bpviol CRC4 quadBRI PCI ISDN Card 1 Span 1 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 2 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 3 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 4 [TE] (caˇ OK 0 0 0 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 and it seems to work. Only sip phones connected to *PBX (by gateways or ethernet) as i remember installation process: 1. bristuff patching asterisk/libpri/zaptel 2. libpri/zaptel/asterisk install 3. zaptel/quozap/wctdm modules installation Runs on Debian 3.1. kernel 2.6.15.4 but i have problem: when SIP hardphone hangup connection (SIP/ - Zap/) asterisk don't send Q.931 DISCONNECT message, and i don't have any idea how to fight with that. Filip D. Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can'tget it to work.-David-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Julian J.M.Sent: Friday, March 31, 2006 1:44 AMTo: Chris Earle; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: BRI cards, HFC,and bristuff - ageneral questionto clear up my understanding.I have 2 different instalations with 1 Billion HFC Card (1port), and 1TDM400. Asterisk 1.0.10+bristuff+florz patch.Only issue is that you must load all modules (wcfxs, zaphfc) beforeruning ztcfg, otherwise nothing works.Everything works ok, even faxing.Julian.On 3/30/06, Chris Earle [EMAIL PROTECTED] wrote: What? After hours of searching for anything to help me, I found thiscomment about zaptel cards in systems with bristuff-cards (junghanns for me in this case)I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine?thanks :-SZoa [EMAIL PROTECTED] wrote in messagenews:[EMAIL PROTECTED]... snip We stopped with the bristuff as bristuff will break any other zaptelcards in the same system. (pri seems logical, why the tdm card alsobroke is unknown to me). snip ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users__This email has been scanned by the MessageLabs Email Security System.For more information please visit http://www.messagelabs.com/email_--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --___Filip DrągowskiMobile: +48(0)500 054045E-mail: [EMAIL PROTECTED]___ONTP.NET Tomasz KarczewskiAleja Wojska Polskiego 33 pokoj 12265-077 Zielona Góra, PolandMobile: +48(0)501 653395Office: +48(0)68 4141018Fax: +48(0)68 4141017http://www.ontp.net___ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Hylafax, on the same box
On Fri, 31 Mar 2006, Olivier Krief wrote: 2006/3/31, Armin Schindler [EMAIL PROTECTED]: Yes, this is possible of course with the Eicon Diva Server PRI (T1) card. This card provides a CAPI interface where you can connect Asterisk(with chan-capi) and any other CAPI based application like Hylafax. You can e.g. configure chan-capi to use 20 channels and the remaining channels configured in Hylafax. When you use a Eicon Diva Server with DSPs on board, you don't need to worry about CPU power, because fax-receive is done on the DIVA card. So you don't need to 'bridge' something here, it just works. Armin Armin, Do you mean you could dynamically share E1/T1 channels between Asterisk and Hylafax applications ? Yes, CAPI provides all available controllers (ports) and its channels to every application at the same time. For example, for each incoming call to a given fax number, Capi driver would trigger Hylafax software to process incoming fax and at the same time, Asterisk software would be smart enough to use other channels for outgoing calls ? Yes, via the CAPI interface you don't reference a real b-channel, this is done by the driver of the ISDN card which provides the CAPI interface. Using CAPI, the applications can (and have to) decide which calls they want to get signaled or which are ignored when they are meant for another service. E.g. the following example is not possible with CAPI: You have one number (and the same BC) for two services assigned. If you are using one application, which can switch to another server by some rule, then it is okay. But two applications must be configured to serve the own numbers/services only. Another thing is, the application does not know about busy channels. This means if you have a 23 channel line and 10 lines are busy with hylafax at the moment, then chan-capi (or another application) can use 13 channels only, of course. So if you have configured chan-capi with e.g. 15 channels to use, chan-capi will just return 'busy : no circuit/channel available'. But this is all configuration stuff and when configured correctly, it works very good. There are even more capabilities. For example Eicon is doing a lot. Their Diva Server cards do provide a RTP interface via CAPI (new chan-capi will support this). Which means coding and anti-jitterbuffer is done on the ISDN card, chan-capi just 'pushes' the RTP packets onto the card... With rcapid and a patched version of the libcapi.so, you can even have the ISDN hardware on one server and the applications on other servers connected via CAPIoverTCP (bintec protocol in that case). I use this because my faxing application (just the capifaxrecvd) runs on my local maschine instead of the ISDN/Asterisk/Gateway server. If this understanding is correct, what is the downside ? Why don't everybody use chan-capi ? CAPI comes originally from the Windows world, but is a common ISDN API standard www.capi.org. So if there would be CAPI drivers for all of these ISDN cards, you can use CAPI (chan-capi). So the missing part is the card-driver. Currently I know of three CAPI based hardware: 1) Eicon Diva Server (all cards including analog ports) with full CAPI 2.0 and VoIP/T.38 extensions. 2) AVM (basic CAPI 2.0) 3) mISDN driver for passive cards (hscx/hfc/...) and on BSD with i4b! Armin (www.chan-capi.org ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Hylafax, on the same box
Armin, Thanks a lot for the very detailed answer. I'll have to take a long look at the CAPI interfaces and see how I can pull all this off, it's all very new to me, but at least I understand that with an Eicon card, I could share a T1 between Asterisk and Hylafax. I'm not clear on whether I could receive fax and voice on the same DID though (if its Fax, Hylafax takes care of it, if its voice, Asterisk does)... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: March 31, 2006 7:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Hylafax, on the same box On Fri, 31 Mar 2006, Olivier Krief wrote: 2006/3/31, Armin Schindler [EMAIL PROTECTED]: Yes, this is possible of course with the Eicon Diva Server PRI (T1) card. This card provides a CAPI interface where you can connect Asterisk(with chan-capi) and any other CAPI based application like Hylafax. You can e.g. configure chan-capi to use 20 channels and the remaining channels configured in Hylafax. When you use a Eicon Diva Server with DSPs on board, you don't need to worry about CPU power, because fax-receive is done on the DIVA card. So you don't need to 'bridge' something here, it just works. Armin Armin, Do you mean you could dynamically share E1/T1 channels between Asterisk and Hylafax applications ? Yes, CAPI provides all available controllers (ports) and its channels to every application at the same time. For example, for each incoming call to a given fax number, Capi driver would trigger Hylafax software to process incoming fax and at the same time, Asterisk software would be smart enough to use other channels for outgoing calls ? Yes, via the CAPI interface you don't reference a real b-channel, this is done by the driver of the ISDN card which provides the CAPI interface. Using CAPI, the applications can (and have to) decide which calls they want to get signaled or which are ignored when they are meant for another service. E.g. the following example is not possible with CAPI: You have one number (and the same BC) for two services assigned. If you are using one application, which can switch to another server by some rule, then it is okay. But two applications must be configured to serve the own numbers/services only. Another thing is, the application does not know about busy channels. This means if you have a 23 channel line and 10 lines are busy with hylafax at the moment, then chan-capi (or another application) can use 13 channels only, of course. So if you have configured chan-capi with e.g. 15 channels to use, chan-capi will just return 'busy : no circuit/channel available'. But this is all configuration stuff and when configured correctly, it works very good. There are even more capabilities. For example Eicon is doing a lot. Their Diva Server cards do provide a RTP interface via CAPI (new chan-capi will support this). Which means coding and anti-jitterbuffer is done on the ISDN card, chan-capi just 'pushes' the RTP packets onto the card... With rcapid and a patched version of the libcapi.so, you can even have the ISDN hardware on one server and the applications on other servers connected via CAPIoverTCP (bintec protocol in that case). I use this because my faxing application (just the capifaxrecvd) runs on my local maschine instead of the ISDN/Asterisk/Gateway server. If this understanding is correct, what is the downside ? Why don't everybody use chan-capi ? CAPI comes originally from the Windows world, but is a common ISDN API standard www.capi.org. So if there would be CAPI drivers for all of these ISDN cards, you can use CAPI (chan-capi). So the missing part is the card-driver. Currently I know of three CAPI based hardware: 1) Eicon Diva Server (all cards including analog ports) with full CAPI 2.0 and VoIP/T.38 extensions. 2) AVM (basic CAPI 2.0) 3) mISDN driver for passive cards (hscx/hfc/...) and on BSD with i4b! Armin (www.chan-capi.org ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting a Grandstream Handytone 486 to Asterisk
Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call doesn't get forwarded to asterisk. I used tcpdump to see whether the Handytone sends data packages at all, but it doesn't. Unfortunately the Grandstream support didn't answer my support request. Does someone of you know how to connect the Handytone to the asterisk server, maybe I need a special cable? Thanks for any hints, Ralf Is the device an FXO or FXS. From what it seems (that you are connecting a phone to it) that it is an FXS. This will not allow inbound calls from a PBX. You need an FXO. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Howto cut the first digit
Title: [Asterisk-Users] Howto cut the first digit
Christian Reelfs wrote:
example:
044612345
should be after cut operation:
44612345
My try in the extension.conf:
exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)
exten = _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)
but it didn't work, my problem is the delemiter, I have no delemiter,
the default is - but how to use the function cut() without an delemiter?
Just snip the first digit of a phonenumber.
Use the substring notation as in:
${mynum:1}
which snips the first character from the string.
See the docs for more info http://www.voip-info.org/wiki/view/Asterisk+variables
Trevor Raynsford
Software Engineer
Aculab
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Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
snip Can Asterisk serve as an access server/gateway to the internet? /snip I have the same question. If I had a PRI coming in to asterisk can I have users dial in and have asterisk work as a gateway to the internet ? Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Marketing Materials
We created out own. You should do the same. --- Bob McDowell [EMAIL PROTECTED] wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX: Auto-congesting call due to slow response
Hi, I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to 'qualify=yes'... Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
It's look like this: incomming connection channel = Zap/7, dstchannel = SIP/200 (SIP hardphone) when SIP/200 hangs up Asterisk do: - Changing state for Zap/7 - state 1 (Not in use) and - Changing state for SIP/200 - state 1 (Not in use) but don't send ISDN-Q.931 DISCONNECT message to finally release channel. astersik get DISCONNECT from network provider Similar is for outgoing connection from SIP/200 to Zap/7 Asterisk (sometimes) send DISCONNECT but rarely. is this a behaviour of confilcts betwwen bristuff-card and zaptel-cards ? Filip D. Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again but only one direction. -David From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Filip Drągowski Sent: Friday, March 31, 2006 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding. Asterisk 1.2.4 + bristuff-0.3.0-PRE-1l + libpri-1.2.2 + zapte-1.2.3 + *CLI zap show status Description Alarms IRQ bpviol CRC4 quadBRI PCI ISDN Card 1 Span 1 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 2 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 3 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 4 [TE] (caˇ OK 0 0 0 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 and it seems to work. Only sip phones connected to *PBX (by gateways or ethernet) as i remember installation process: 1. bristuff patching asterisk/libpri/zaptel 2. libpri/zaptel/asterisk install 3. zaptel/quozap/wctdm modules installation Runs on Debian 3.1. kernel 2.6.15.4 but i have problem: when SIP hardphone hangup connection (SIP/ - Zap/) asterisk don't send Q.931 DISCONNECT message, and i don't have any idea how to fight with that. Filip D. Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC,and bristuff - a general questionto clear up my understanding. I have 2 different instalations with 1 Billion HFC Card (1port), and 1 TDM400. Asterisk 1.0.10+bristuff+florz patch. Only issue is that you must load all modules (wcfxs, zaphfc) before runing ztcfg, otherwise nothing works. Everything works ok, even faxing. Julian. On 3/30/06, Chris Earle [EMAIL PROTECTED] wrote: What? After hours of searching for anything to help me, I found this comment about zaptel cards in systems with bristuff-cards (junghanns for me in this case) I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine? thanks :-S "Zoa" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... snip We stopped with the bristuff as bristuff will break any other zaptel cards in the same system. (pri seems logical, why the tdm card also broke is unknown to me). snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email
Re: [Asterisk-Users] Calling home while on the road, will it work?
Yes it will work but you have to set up the dial plan to do what you want it to do. What exactly do you want to happen when you use your soft phone ? --- Kiffin Gish [EMAIL PROTECTED] wrote: I have a Digium TDM400P card with 1 FXS and 1 FXO module running on my FreeBSD 6.0 server. While I am on the road, I would like to save on costs by using a soft-phone from my laptop to call in to a telephone connected to this card. I installed both Asterisk and Zaptel drivers from the ports, but still haven't done anything with the configuration files. What else do I require, and what is the mimimum amount of work to get this up and running? Thanks a lot in adavnce. -- Kiffin Rex Gish Gouda, The Netherlands ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple auto attendants
Did you forget to nsert shamelessPlug ? --- Melcon Moraes [EMAIL PROTECTED] wrote: Sure! In fact, there's a nice GUI for setting up all this, called Phonecall. Check it out in http://www.vecsector.com/phonecall/ You can do it all by your hands as well. :) []'s MM -Original Message- From: Paolo Supino [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Thu, 30 Mar 2006 17:03:15 -0500 Delivered: Thu, 30 Mar 2006 19:01:02 Subject:[Asterisk-Users] multiple auto attendants Hi I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? TIA Paolo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1143756423.564653.8656.mangoro.hst.terra.com.br,3635,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo still present with Eicon Diva Server 4 Bri
Hi! I'm trying to enable echo cancelation with my Eicon Diva Server 4 Bri card. I've enabled it from zapata.conf, (as I read from www.voip-info.org) --- ; Enable echo cancellation echocancel=64 echocancelwhenbridged=yes echotraining=2000 but nothing seems to change, echo is still present. Any idea? Should I enable/change something else? Is there a way from console to see if echo cancelation is active? Thanks for your time! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode
if the attachment has something else, please forward it to eaperezh (at) gmail (dot) com thanks for the info. On 3/31/06, artifex maximus [EMAIL PROTECTED] wrote: I had conversation with Welltech support and I got this description (I can't send attachment through the list): The 380x has a routing table function. There are two default route exist in the routing table, one is for IP incoming call another is for FXO incoming call, the IP call will be routed to FXO and the Call from FXO side will be routed to IP side. If the 380x got an incoming call from IP side and the line number is not it's local SIP number. Then the 380x will forward this number to the FXO which is based on the default route, and the FXO will dial this number through PSTN automatically. User just need to dial the destination PSTN number, this is called one-stage dialing. For more information about one-stage dialing, please refer to the attachment. If the 380x got an incoming call from IP side, and the line number is the FXO's local SIP number, then the FXO will answer this call and user will hear dial tone from PSTN side, then they should re-dial the destination PSTN number. User should dial number twice, so this is called two-stage dialing. artifex On 3/31/06, Erick Perez [EMAIL PROTECTED] wrote: one-stage calling function? On 3/30/06, kevin ling [EMAIL PROTECTED] wrote: Yes, Same configuration as Martin. 1.for incoming call just set the 3804 hotline to one sip extension number. 2.for outgoing call, you just using regular dail command to pass the phone number to 3804 (3804 is a 4FXO port device, the call from ip side always pass to FXO Port). You can telnet to the 3804 and enable the one-stage calling function. Regards, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX: Auto-congesting call due to slow response
I have same problem, do you have asterisk box behind nat? PJ Mimmus wrote: Hi, I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to 'qualify=yes'... Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How is Teliax ?
[EMAIL PROTECTED] wrote: Lonnie Abelbeck wrote: asterisk at anime.net writes: On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote: I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers. I have had issues with termination on teliax. Callers tell me I sound choppy to them. Teliax origination has no problems at all strangely enough. If you used SIP instead of IAX2 with Teliax you will have better quality calls. The 'choppy' sound occurs with IAX2 and not SIP at Teliax. I can recommend Teliax, but use SIP. But I _am_ using SIP. I tried all the various teliax gateways including the beta test ones and had choppiness with all of them. As I said before, teliax origination had no choppiness problems at all. Only termination had issues. I had no problems - termination or origination - with junction networks, despite the fact they had 3x higher latency than teliax. JN is more expensive than teliax though. Also, I have talked to others who had similar choppiness problems with teliax. So it's not just me. For whatever its worth, the majority of teliax users I'm sure have had at least some audio choppiness. Teliax is obviously aware of this since they have spent a fair amount of time recently moving/enhancing their implementations. It probably has nothing to due with iax vs sip other then those around this list know that at least some changes/improvements have been also occurring with the iax code. One other consideration is that its obvious calls are handled in different ways depending on the origination and termination points. In other words, various npa-nxx calls are handed off to different wholesale providers that can also be the source of call quality issues. (I've identified some very specific npa's where this happens to be the case.) There are a couple of other well known itsp's that don't have those issues, and its highly likely their implementations in terms of wholesale delivery sources are different. Overall, I'd give teliax folks high marks for paying attention to customer service and addressing issues, even though they are very closed-mouth about service improvement plans, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX: Auto-congesting call due to slow response
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek I have same problem, do you have asterisk box behind nat? No, they are not behind NAT, peraphs there is a Checkpoint firewall. DV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Hylafax, on the same box
On Fri, 31 Mar 2006, Mike wrote: Armin, Thanks a lot for the very detailed answer. I'll have to take a long look at the CAPI interfaces and see how I can pull all this off, it's all very new to me, but at least I understand that with an Eicon card, I could share a T1 between Asterisk and Hylafax. I'm not clear on whether I could receive fax and voice on the same DID though (if its Fax, Hylafax takes care of it, if its voice, Asterisk does)... Fax and voice on the same DID is not possible when using a second application like hylafax. Because how should the two applications decide which one accepts the call? But you can receive faxes with chan-capi (when you have an Eicon can with DSPs which does fax-processing on board). Here you can use the dialplan to decide what to do with the call. I do this in a company (OpenPBX in that case, but it's the same), I receive faxes via CAPI and sending is done with another application. Armin Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: March 31, 2006 7:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Hylafax, on the same box On Fri, 31 Mar 2006, Olivier Krief wrote: 2006/3/31, Armin Schindler [EMAIL PROTECTED]: Yes, this is possible of course with the Eicon Diva Server PRI (T1) card. This card provides a CAPI interface where you can connect Asterisk(with chan-capi) and any other CAPI based application like Hylafax. You can e.g. configure chan-capi to use 20 channels and the remaining channels configured in Hylafax. When you use a Eicon Diva Server with DSPs on board, you don't need to worry about CPU power, because fax-receive is done on the DIVA card. So you don't need to 'bridge' something here, it just works. Armin Armin, Do you mean you could dynamically share E1/T1 channels between Asterisk and Hylafax applications ? Yes, CAPI provides all available controllers (ports) and its channels to every application at the same time. For example, for each incoming call to a given fax number, Capi driver would trigger Hylafax software to process incoming fax and at the same time, Asterisk software would be smart enough to use other channels for outgoing calls ? Yes, via the CAPI interface you don't reference a real b-channel, this is done by the driver of the ISDN card which provides the CAPI interface. Using CAPI, the applications can (and have to) decide which calls they want to get signaled or which are ignored when they are meant for another service. E.g. the following example is not possible with CAPI: You have one number (and the same BC) for two services assigned. If you are using one application, which can switch to another server by some rule, then it is okay. But two applications must be configured to serve the own numbers/services only. Another thing is, the application does not know about busy channels. This means if you have a 23 channel line and 10 lines are busy with hylafax at the moment, then chan-capi (or another application) can use 13 channels only, of course. So if you have configured chan-capi with e.g. 15 channels to use, chan-capi will just return 'busy : no circuit/channel available'. But this is all configuration stuff and when configured correctly, it works very good. There are even more capabilities. For example Eicon is doing a lot. Their Diva Server cards do provide a RTP interface via CAPI (new chan-capi will support this). Which means coding and anti-jitterbuffer is done on the ISDN card, chan-capi just 'pushes' the RTP packets onto the card... With rcapid and a patched version of the libcapi.so, you can even have the ISDN hardware on one server and the applications on other servers connected via CAPIoverTCP (bintec protocol in that case). I use this because my faxing application (just the capifaxrecvd) runs on my local maschine instead of the ISDN/Asterisk/Gateway server. If this understanding is correct, what is the downside ? Why don't everybody use chan-capi ? CAPI comes originally from the Windows world, but is a common ISDN API standard www.capi.org. So if there would be CAPI drivers for all of these ISDN cards, you can use CAPI (chan-capi). So the missing part is the card-driver. Currently I know of three CAPI based hardware: 1) Eicon Diva Server (all cards including analog ports) with full CAPI 2.0 and VoIP/T.38 extensions. 2) AVM (basic CAPI 2.0) 3) mISDN driver for passive cards (hscx/hfc/...) and on BSD with i4b! Armin (www.chan-capi.org ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
Don Pobanz wrote: Adolfo R. Brandes wrote: Lee Howard wrote: However, based on the comments you give I'd suspect that you're having what people seem to be calling frame slipping. There seem to be some motherboards that react poorly with Zap cards (or their respective drivers) and cause that. Your zttest results should be revealing here. Frame slips are NOT motherboard related! A Frame slip is due to clocks at opposite ends of a circuit such as a T1 running at different speeds. Either a buffer overflows and one frame is thrown away or there is no data when a frame is needed so the previous frame is repeated. The solution is to have one end of the circuit supply the clock and the other end derive the clock from the incoming signal. True frame slips are not motherboard related. However, many people loose samples (usually chunks of 8 or 160) due to motherboard (or possibly BIOS) issues. Some motherboards seem far more prone than others to loosing interrupts at the high rate these boards work. That might be to do with the PCI latency settings, or PCI controller effeciency or a bunch of other variables. However, the bottom line is people do loose samples due to motherboard issues, and frame slips, not entirely unreasonably, tends to get used as a catch all term for these things. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
Appreciate the replies everyone -- really I'm wondering if I should be using zapHFC with my Junghanns card instead of qozap? Everyone always mentions zaphfc -- mostly I guessed because they are using a zaphfc-compatible card - but *maybe* I should try that instead of qozap??? And yep -- totally know about the module load order thing and ztcfg -- no worries there I've been able to dial out and everything from the start ! -- which is a bridge from digium--junghanns there..but incoming calls seem to be a whole other issue. :-( Exhausted from trying a million things, Chris Chris Earle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] What? After hours of searching for anything to help me, I found this comment about zaptel cards in systems with bristuff-cards (junghanns for me in this case) I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine? thanks :-S Zoa [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] snip We stopped with the bristuff as bristuff will break any other zaptel cards in the same system. (pri seems logical, why the tdm card also broke is unknown to me). snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Hylafax, on the same box
That's not entirely correct :) Fax and voice on the same DID is not possible when using a second application like hylafax. Because how should the two applications decide which one accepts the call? With the help of iaxmodem (which works really well) its easily done! Just detect the incoming call is fax and the route it to iaxmodem on fax extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo still present with Eicon Diva Server 4 Bri
When using Eicon Diva Server, you use chan-capi! And the zapata.conf is not read by chan-capi. You need to setup capi.conf. See the example and the README provided by chan-capi package from chan-capi.org Armin On Fri, 31 Mar 2006, Giuseppe wrote: Hi! I'm trying to enable echo cancelation with my Eicon Diva Server 4 Bri card. I've enabled it from zapata.conf, (as I read from www.voip-info.org) --- ; Enable echo cancellation echocancel=64 echocancelwhenbridged=yes echotraining=2000 but nothing seems to change, echo is still present. Any idea? Should I enable/change something else? Is there a way from console to see if echo cancelation is active? Thanks for your time! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I have debug off why are the logs show debug info
Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:35 DEBUG[2423] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Mar 31 06:57:12 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Building Asterisk embedded device
Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Thanks A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Building Asterisk embedded device
Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Thanks A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
However, based on the comments you give I'd suspect that you're having what people seem to be calling frame slipping. There seem to be some motherboards that react poorly with Zap cards (or their respective drivers) and cause that. Your zttest results should be revealing here. Frame slips are NOT motherboard related! For those that happen to be following this thread, be careful with the above statement as one person is assuming T1/E1 interfaces while another is assuming the statement applies to analog interfaces such as the TDM400 card. The statement applies to one assumption but not the other. A Frame slip is due to clocks at opposite ends of a circuit such as a T1 running at different speeds. Either a buffer overflows and one frame is thrown away or there is no data when a frame is needed so the previous frame is repeated. The solution is to have one end of the circuit supply the clock and the other end derive the clock from the incoming signal. Don Pobanz How would you check clocks speeds at opposite ends of a circuit (T1, E1, BRI, ...) ? As it seems frame slips occur from time to time (for instance, on 10% of received faxes), do you imply that Asterisk settings should be changed so that on every fax received, it should adopt opposite clock speed (unlike today where by chance, 90% of circuit clock speeds are the same) ? The issue on T1/E1's is not clock speed itself but rather the low level synchronization of the clocks. The timing signals necessary for clock synchronization are always embedded in the transmit side of every T1/E1 data stream. Its part of the T1/E1 design specifications and cannot be removed under any circumstance by anyone. Whether you use it or not is defined in /etc/zaptel.conf. The telco's and other T1/E1 service providers never listen to your synchronization; rather, they have a very well understood hierarchy where synchronization is always derived from their upstream providers (whoever they happens to be). Therefore, if you synchronize your clocks to the T1/E1 provided to you, your clocks will be synchronized to the rest of the world (in total). FWIW, the folks at sangoma have said that one of there design verification tests includes ensuring that faxing works since it is one of the most critical tests that validates overall design. So, if you are absolutely sure that you've specified the correct T1 synchronization parameters in your /etc/zaptel.conf and you still have fax reliability issues, look elsewhere in your implementation for the root cause. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I have debug off why are the logs show debug info
Look what you have in /etc/asterisk/logger.conf find: console = message = full = Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:35 DEBUG[2423] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Mar 31 06:57:12 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Filip Drągowski Mobile: +48(0)500 054045 E-mail: [EMAIL PROTECTED] ___ ONTP.NET Tomasz Karczewski ul. Cynarskiego 5 65-831 Zielona Góra, Poland Mobile: +48(0)501 653395 Office: +48(0)68 4141018 Fax:+48(0)68 4141017 http://www.ontp.net ___ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multicast Music on Hold
Nathan Alberti wrote: As i am using a central asterisk box with multiple stub sites I don't wish every call put on hold to be wasting WAN bandwidth, I am wondering if it is possible to create a multicast stream to each site and rather than asterisk sending its address and the media information during a hold it sends the multicast address and multiple phones can be served by the one stream ? This is certainly possible, but I doubt that most existing SIP phones have the ability to subscribe to multicast groups and handle it properly. Without that, it would only work on a LAN (non-routed). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Building Asterisk embedded device
On 31/03/06, sam [EMAIL PROTECTED] wrote: Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Depends what horsepower you'll need - many people have had good results with the Soekris NET4801, running Astlinux. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 - unable to install
I'm and [EMAIL PROTECTED] user - been so now for almost a year. Lately, I've upgraded to the latest greatest.. (which is built on 1.2.5) and am unable to install oh323. I've already asked over at the ([EMAIL PROTECTED]) Sourceforge forum but no one seems to think it worth answering. The error I get is pretty obvious but I don't know where to go from here. More importantly, I need to have a workable solution - My question to the [EMAIL PROTECTED] gang was whether oh323 works (before I actually tried to install it). Here's the error (or where it starts). vpwlib/ChangeLog checking for g++... no checking for c++... no checking for gpp... no checking for aCC... no checking for CC... no checking for cxx... no checking for cc++... no checking for cl... no checking for FCC... no checking for KCC... no checking for RCC... no checking for xlC_r... no checking for xlC... no checking for C++ compiler default output... configure: error: C++ compiler cannot create executables See `config.log' for more details. make: *** No rule to make target `clean'. Stop. make: *** No rule to make target `opt'. Stop. checking for g++... no checking for c++... no checking for gpp... no checking for aCC... no checking for CC... no checking for cxx... no checking for cc++... no checking for cl... no checking for FCC... no checking for KCC... no checking for RCC... no checking for xlC_r... no checking for xlC... no checking for C++ compiler default output... configure: error: C++ compiler cannot create execu tables See `config.log' for more details. make: *** No rule to make target `clean'. Stop. make: *** No rule to make target `opt'. Stop. for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.5/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323V ERSION=\1.13.5\ -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/ope nh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o make[1]: g++: Command not found make[1]: *** [wrapper_misc.o] Error 127 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.5/wrapper' make: *** [subdirs_build] Error 1 for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.5/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323V ERSION=\1.13.5\ -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/ope nh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o make[1]: g++: Command not found make[1]: *** [wrapper_misc.o] Error 127 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.5/wrapper' make: *** [subdirs_build] Error 1 --- Installing GnuGK --- cp: cannot create regular file `/usr/sbin/gnugk': Text file busy mkdir: cannot create directory `/var/log/gk/': File exists STOPPING ASTERISK -- Remote UNIX connection Disconnected from Asterisk server Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started H.323 support installed. BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD
Re: [Asterisk-Users] Building Asterisk embedded device
On 00:01, Sat 01 Apr 06, sam wrote: Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. http://www.soekris.com -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Building Asterisk embedded device
http://gumstix.com/waysmalls.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sam Sent: Friday, March 31, 2006 8:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Building Asterisk embedded device Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Thanks A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check if a phone / line is used?
Show channels? On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote: In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? Sip show peers shows me just if it is on-line, but not if it is in a call or not. In the dialplan I could dial the number and if it is busy, it would go to the Voicemail for unavailable or busy. I expect that there is just a test function as well, without trying to call. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How is Teliax ?
You cannot criticize Teliax until you investigate how your calls are getting to them. I have a customer on 17th St. in downtown Denver who use Qwest.net as their ISP. They use Teliax (on 16th St.) as their ITSP. Piece of cake, right? This may have changed recently, but Qwest doesn't have any peering arrangements in Denver (!), so, to get to the rest of the Internet, Qwest traffic is routed over a very congested circuit to Dallas where it has a peering arrangement with Sprint. For the ordinary Internet user, this trip to Dallas using TCP won't be noticed. The TCP protocol will resend any dropped packets. For my VoIP customer having UDP packets dropped at congested routers in Dallas, it's a disaster. This VoIP connection between Qwest.net and Teliax is not suitable for VoIP, and it's not the fault of Teliax. Having said all that, I see where Teliax have installed the voip-co4 host on Viawest. Are you using that host for your analysis? [EMAIL PROTECTED] wrote: Lonnie Abelbeck wrote: asterisk at anime.net writes: On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote: I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers. I have had issues with termination on teliax. Callers tell me I sound choppy to them. Teliax origination has no problems at all strangely enough. If you used SIP instead of IAX2 with Teliax you will have better quality calls. The 'choppy' sound occurs with IAX2 and not SIP at Teliax. I can recommend Teliax, but use SIP. But I _am_ using SIP. I tried all the various teliax gateways including the beta test ones and had choppiness with all of them. As I said before, teliax origination had no choppiness problems at all. Only termination had issues. I had no problems - termination or origination - with junction networks, despite the fact they had 3x higher latency than teliax. JN is more expensive than teliax though. Also, I have talked to others who had similar choppiness problems with teliax. So it's not just me. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How is Teliax ?
We use them for origination over IAX. At first we had callee's reporting that our voice was choppy to them, while the callee has always sounded fine on our end. I made that problem go away by introducing traffic shaping at our firewall. They have a bug, whereby on their website you can set the codecs allowed to be used by your account, on both SIP and IAX, but that the settings to do stick. So, say, you are trying to setup your account, and you tell TelIAX not to allow ulaw. Maybe later on you do want to use ulaw, so you set your account at TelIAX to allow it, but it doesn't effect any change. ...in order to fix it you must hound their techs for a few days to reset your account. MORAL OF THE STORY: ALWAYS LEAVE EVERY CODEC _ON_ AT THE TELIAX WEBSITE. USE YOUR IAX.CONF/SIP.CONF TO NEGOTIATE THE CODEC. I recommend them, just not their codec selection website interface, which an asterisk user probably doesn't need anyways. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hosted solution
http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I have debug off why are the logs show debug info
Hi, Thank you that was it, I had 'debug' listed under 'full' in logger.conf. Not sure how I missed that... Thanks Again Filip Drągowski wrote: Look what you have in /etc/asterisk/logger.conf find: console = message = full = Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:35 DEBUG[2423] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Mar 31 06:57:12 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXY codec support and questions..
Hi..
I have to setup an extension in a remote location that will use a
cordless analog telephone.. I am looking at the IAXY to do this for
me..Basically the data path will be as follows...
[Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset
Since there are two NAT boxes in the path I know SIP won't work.. I also
don't want to move the Asterisk box to the internet side of the NAT box,
not only from the security perspective but also the potential issues
with the already configured SIP phones that connect to it locally..
So as far as getting over the NAT problem the IAXY seems the way to go..
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
Thanks for any suggestions or input on this setup.. Also any reviews on
the IAXY are welcome..
___
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[Asterisk-Users] meetme option 'e'
Hi, Option 'e' is for selecting an empty conference to join. My question is. How do I know what the conference number is for the next party to join? Does it set it to a variable? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Hello Dinesh I got a Panasonic KX-TDA100, can you tell me please how can you configure the PBX side? Qsig slave? master? and the other side of the asterisk? I got TE100P Regards, Daniel Dinesh Nair wrote: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXY codec support and questions..
WipeOut wrote: To save bandwidth I would like to stay away from using the G.711 codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in the docs.. No. The IAXy only supports G.711 ulaw/alaw and ADPCM. I don't know what 'docs' you were looking in, but this page: http://www.digium.com/en/docs/S101I/IAXy.pdf clearly states what is supported. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cannot set outgoing cid
(macro-record-enable,s,4) Mar 31 16:53:56 DEBUG[11747] pbx.c: Launching 'AGI' Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Executing AGI(SIP/451-0e31, recordingcheck|20060331-165356|1143816836.643) in new stack Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Mar 31 16:53:57 VERBOSE[11747] logger.c: recordingcheck|20060331-165356| 1143816836.643: Outbound recording not enabled Mar 31 16:53:57 VERBOSE[11747] logger.c: -- AGI Script recordingcheck completed, returning 0 Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'NoOp' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing NoOp(SIP/451-0e31, No recording needed) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'Macro' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing Macro(SIP/451-0e31, outbound-callerid|1) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'DBget' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing DBget(SIP/451-0e31, USEROUTCID=AMPUSER/451/outboundcid) in new stack Mar 31 16:53:57 VERBOSE[11747] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=451/outboundcid Mar 31 16:53:57 VERBOSE[11747] logger.c: -- DBget: set variable USEROUTCID to 033811234451 Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?4) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetCallerID' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetCallerID(SIP/451-0e31, 033811234100) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?6) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetCallerID' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetCallerID(SIP/451-0e31, 033811234451) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'NoOp' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing NoOp(SIP/451-0e31, CallerID set to 033811234451) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetGroup' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetGroup(SIP/451-0e31, OUT_1) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Function result is '1' Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?108) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetVar' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetVar(SIP/451-0e31, DIAL_NUMBER=03381765432) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetVar' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetVar(SIP/451-0e31, DIAL_TRUNK=1) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'AGI' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing AGI(SIP/451-0e31, fixlocalprefix) in new stack Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Mar 31 16:53:57 VERBOSE[11747] logger.c: fixlocalprefix: Removed prefix. New number: 3381765432 Mar 31 16:53:57 VERBOSE[11747] logger.c: -- AGI Script fixlocalprefix completed, returning 0 Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetVar' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetVar(SIP/451-0e31, OUTNUM=3381765432) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'Cut' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing Cut(SIP/451-0e31, custom=OUT_1|:|1) in new stack Mar 31 16:53:57 WARNING[11747] ast_expr2.y: non-numeric argument Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?16) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'Dial' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing Dial(SIP/451-0e31, ZAP/g1/3381765432) in new stack Mar 31 16:53:57 DEBUG[11747] chan_zap.c: Using channel 1 Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable STACK-macro-dialout-trunk-s-14. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable MACRO_DEPTH. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable STACK-macro-dialout-trunk-s-13. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable custom. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable STACK-macro-dialout-trunk-s-12. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable OUTNUM. Mar
[Asterisk-Users] decrease the speed of reading text!!!
Hi all How can i decrease the speed of festival? It appear that in festival, the text is read too fast for me ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot set outgoing cid
' Mar 31 16:53:56 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0 0?2:4) in new stack Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Goto (macro-record-enable,s,4) Mar 31 16:53:56 DEBUG[11747] pbx.c: Launching 'AGI' Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Executing AGI(SIP/451-0e31, recordingcheck|20060331-165356|1143816836.643) in new stack Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Mar 31 16:53:57 VERBOSE[11747] logger.c: recordingcheck|20060331-165356| 1143816836.643: Outbound recording not enabled Mar 31 16:53:57 VERBOSE[11747] logger.c: -- AGI Script recordingcheck completed, returning 0 Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'NoOp' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing NoOp(SIP/451-0e31, No recording needed) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'Macro' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing Macro(SIP/451-0e31, outbound-callerid|1) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'DBget' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing DBget(SIP/451-0e31, USEROUTCID=AMPUSER/451/outboundcid) in new stack Mar 31 16:53:57 VERBOSE[11747] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=451/outboundcid Mar 31 16:53:57 VERBOSE[11747] logger.c: -- DBget: set variable USEROUTCID to 033811234451 Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?4) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetCallerID' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetCallerID(SIP/451-0e31, 033811234100) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?6) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetCallerID' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetCallerID(SIP/451-0e31, 033811234451) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'NoOp' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing NoOp(SIP/451-0e31, CallerID set to 033811234451) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetGroup' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetGroup(SIP/451-0e31, OUT_1) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Function result is '1' Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?108) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetVar' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetVar(SIP/451-0e31, DIAL_NUMBER=03381765432) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetVar' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetVar(SIP/451-0e31, DIAL_TRUNK=1) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'AGI' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing AGI(SIP/451-0e31, fixlocalprefix) in new stack Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Mar 31 16:53:57 VERBOSE[11747] logger.c: fixlocalprefix: Removed prefix. New number: 3381765432 Mar 31 16:53:57 VERBOSE[11747] logger.c: -- AGI Script fixlocalprefix completed, returning 0 Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'SetVar' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing SetVar(SIP/451-0e31, OUTNUM=3381765432) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'Cut' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing Cut(SIP/451-0e31, custom=OUT_1|:|1) in new stack Mar 31 16:53:57 WARNING[11747] ast_expr2.y: non-numeric argument Mar 31 16:53:57 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?16) in new stack Mar 31 16:53:57 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16:53:57 DEBUG[11747] pbx.c: Launching 'Dial' Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Executing Dial(SIP/451-0e31, ZAP/g1/3381765432) in new stack Mar 31 16:53:57 DEBUG[11747] chan_zap.c: Using channel 1 Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable STACK-macro-dialout-trunk-s-14. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable MACRO_DEPTH. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying
RE: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Looking at the TE100P I don't see it listed Q.SIG as supported. We have it working great as PRI. Am I wrong about the Q.SIG support? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Sent: Friday, March 31, 2006 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX? Hello Dinesh I got a Panasonic KX-TDA100, can you tell me please how can you configure the PBX side? Qsig slave? master? and the other side of the asterisk? I got TE100P Regards, Daniel Dinesh Nair wrote: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot set outgoing cid
Hi Sebastian - sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my extension. any hint to what could be wrong is greatly appreciated. There's no greeat trick to it. Your provider just has to support it. My PRI provider here in the US allows me to set my CID, but it has to fall exactly within certain parameters: it has to be 9 digits and within my DID block. If it doesn't fit those parameters, the provider will just blank out the caller ID. As I understand it, the policies of various providers vary greatly, and some providers won't allow you to set CID at all. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXY codec support and questions..
Kevin P. Fleming wrote: WipeOut wrote: To save bandwidth I would like to stay away from using the G.711 codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in the docs.. No. The IAXy only supports G.711 ulaw/alaw and ADPCM. I don't know what 'docs' you were looking in, but this page: http://www.digium.com/en/docs/S101I/IAXy.pdf clearly states what is supported. Thanks, I was looking at the install guide.. It may have been in there too and I just missed it.. Unfortunately G.711 is not going to help me.. I could still get by using it but the quality may be an issue when there is other traffic on the line.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
So, if you are absolutely sure that you've specified the correct T1synchronization parameters in your /etc/zaptel.conf and you still have fax reliability issues, look elsewhere in your implementation for theroot cause.So, would you conclude that it's possible for a given T1/E1 to have incorrect T1/E1synchronization parameters and still work 90% of the time ? In other words, if 90% of faxes are correctly received, does it implies I have to look elsewhere in my implementation for the root cause of the remaining 10% ?Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] statechange_queue
Hi, Sometimes my Asterisk displays the following error message... Mar 31 12:53:04 WARNING[17170]: app_queue.c:519 statechange_queue: Failed to create update thread! Has anybody seen it before? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transcoding g723 or g729 on asterisk
Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I dont have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up From [cisco (g729)] [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] - [pstn] And vice versa. I will be glad if someone can throw more light on this for me. Goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Monday, March 27, 2006 3:19 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] g723 or g729 on ss7 link Hi! Is it possible to set codecs on ss7 link? No. E1s channels (which chan_ss7 uses as voice channels) can only use G711 alaw. Or receiving call with g723 or g729 and forward the call to pstn via the ss7 link. You can do transcoding on the asterisk machine that you use as SS7 Gateway. This means you get the calls delivered via SIP with G723 and send them out to SS7 with G711. Note that you need more processingpower for this then if you do no transcoding. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-14 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Dinesh Nair [EMAIL PROTECTED] writes: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. Well, that's the YMMV. I have it also running with an Alcatel 4200. But my last experience with the 4400 showed me that there is something missing in the Q.SIG implementation. I also have seen some weird things with Q.SIG on BRI. And as long as I don't know what will happen when I connect * to some PBX, I won't tell my customers about Q.SIG. cu, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Polycom IP501 and Speed Dials
Hi Avi - I know this is off-topic for Asterisk, but I don't know where else to ask: I've setup a central directory.xml file for my Polycom IP501 phones with a list of all the internal extensions. None of them have sd1/sd as I don't want to enable any speed dials, just have a list in each phone. However, when a phone boots, it seems to pick a random entry and put it on the second line key as a speed dial entry! Anyone have any idea why and how to stop it? Also, could someone confirm that once a phone loads the default directory, it then maintains its own copy? So if I want to change the directory from the FTP server, I have to edit every single phone-specific XML file, or will the phone overwrite that on reboot? Essentially, I'm looking for a way to manage the directory from a central location. I think you may be stuck with the central directory storage. If I remember right, my experience was like yours - you can create a single central directory, and then the phones will copy it to their own individual directory file (mac address-directory.xml). I guess you could delete all the individual directories to force the phones to go back to the central file (I haven't tested, though, so I don't know if this would really work). Kind of a pain, especially since you have to reboot all the phones for this to happen. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo cancellation problem
Hi! I'm here again with echo canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1regID=4999 Card features: * Supplementary Services o Number Identification services (CLIP, CLIR, COLP, COLR, KEY, MSN, DDI, SUB) o Call offering services (TP, CFU, CFB, CFNR) o Call completion services (CW, HOLD, ECT) o Charging services (AoC) o Three-party conference o Large Conference * VoIP Gateway support o Echo cancellation (G.168) 128ms (32ms for VoIP) it seems to support echo cancellation o Real Time Protocol (RTP framing) o Dynamic Anti-Jitter Buffer o Comfort Noise Generation o Voice Activity Detection o Voice compression (GSM, G.726) = capi.conf = [general] nationalprefix=039 ;;internationalprefix=011 rxgain=1.0 txgain=2.0 ;rxgain=1.0 ;txgain=1.0 alaw=yes ; ;ulaw=yes ;set this, if you live in u-law world instead of a-law --beppe [ISDN3] ntmode=yes incomingmsn=* controller=3 group=1 ;softdtmf=on ;relaxdtmf=on ;accountcode= context=isdn3in holdtype=local immediate=yes ;echosquelch=1 echocancel=yes ;echocancelold=yes echotail=64 bridge=yes callgroup=1 ;deflect=1234567 devices=2 == and this is a piece of what asterisk -rv says when someone calls == == ISDN3: Incoming call 'yy' - 'x' -- Executing Set(CAPI/ISDN3/'x'-1, LANGUAGE()=it) in new stack -- Executing GotoIfTime(CAPI/ISDN3/'x'-1, 09:00-13:00|mon-fri|*|*?coda_lib_uni|s|1) in new stack -- Executing GotoIfTime(CAPI/ISDN3/'x'-1, 15:00-18:00|mon-fri|*|*?coda_lib_uni|s|1) in new stack -- Goto (coda_lib_uni,s,1) -- Executing Set(CAPI/ISDN3/'x'-1, LANGUAGE()=it) in new stack -- Executing SetVar(CAPI/ISDN3/'x'-1, MONITOR_FILENAME=lib-uni-31032006-16:40:21-039564615724) in new stack -- Executing Playback(CAPI/ISDN3/'x'-1, wsa_benvenuto_lib_uni) in new stack == ISDN3: Answering for 'x' -- Playing 'wsa_benvenuto_lib_uni' (language 'it') Mar 31 16:40:21 WARNING[30181]: file.c:1029 ast_waitstream: Unexpected control subclass '14' == ISDN3: Setting up echo canceller (PLCI=0x103, function=1, options=4, tail=64) == ISDN3: Setting up DTMF detector (PLCI=0x103, flag=1) -- ISDN3: Error setting up echo canceller (PLCI=0x103) Mar 31 16:40:21 WARNING[29878]: chan_capi.c:3334 show_capi_conf_error: ISDN3: conf_error 0x300b PLCI=0x103 Command=FACILITY_CONF,0x8497 CAPI INFO 0x300b: Facility not supported ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Hylafax, on the same box
On Fri, 31 Mar 2006, Boris Bakchiev wrote: That's not entirely correct :) Fax and voice on the same DID is not possible when using a second application like hylafax. Because how should the two applications decide which one accepts the call? With the help of iaxmodem (which works really well) its easily done! Just detect the incoming call is fax and the route it to iaxmodem on fax extension. Yes, of course, but that wasn't the question. We are talking about two capi applications here, which are completely separated here. Another capi tool could this as well, just accept the call and route it back via another interface to another application. But anyway, it is not necessary at all. chan-capi is doing this with Asterisk/OpenPBX without further tools. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reporting?
NICE! On 3/30/06, Joe Dennick [EMAIL PROTECTED] wrote: I see (and like) the demo, but where can we get it? Doug Lytle wrote: Nicolás Gudiño wrote: shameless plug Something like this perhaps? http://www.asternic.org/stats/demo O VERY cool! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk turn key solution
can anyone recommend a asterisk turn key company. we will need the hardware as well as tech. support 24/7. we'll want all the goodies, voice mail, auto attendant. we have 6 incoming pot lines (all the same number), and 40 normal telephones. we have no interest in changing to ip phones or the pot lines at this time. we're interested in removing our meridian pbx system, installing asterisk, learning how to use it on our own (eventually) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Users/Peers/Friends - Ick
Doug this is in no way an offense to you but I think we need to start the asterisk booze fund. This will be for all of us that have ups and downs in working on getting asterisk set up. I for one have my friend Johny Walker right by my side when ever it gets to me. I'll second that. Maybe not with Johnnie Walker, but seriously, Doug, you're going to give yourself a heart attack worrying about all your asterisk issues. When this stuff gets to me, I gotta realize that no technical issue is ever worth stressing over so much (unless of course, the asterisk box dies at 10:00am on a Monday). Sometimes I get up and take a walk to refresh my perspective a little. For me, at least, answers come much easier when I'm NOT so stressed. All in a day's work. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phones
Olivier Krief wrote: 2006/3/29, John Novack [EMAIL PROTECTED]: The reality is, of course, that telephone systems have provided this function for many years. A DSS/BLF is available on MANY so called legacy systems, so until this function is readily available , customers that require a receptionist will continue to go elsewhere. Perhaps it is time to rethink the way data is exchanged between the CPU and the DSS/BLF? As someone said a very long time ago: Results, not excuses. With user count growing, I think receptionist could evolve from hardware to hardware-software combination the same as receptionist job changes from assisting call transfers (check if someone is available before transfer) to blind call transfering (forward anyway and take the call back if nobody answers). If my understanding is correct, in the later case, a receptionist doesn't really need BLF : he or she simply forward the call. He or she mainly needs a directory application helping him or her to find the right person within the organisation. And I don't think anyone could have the patience to harphone BLF labels every 2 weeks to keep up large site permanent user moves, adds and changes. So the perfect receptionist application hardware-software combination should include a mix between directory application and softphone, and provide comfortable hardware to support these. My opinion is I don't think market trends are at works now to make this perfect combination happen anytime soon : - from my point of view, it could take years to gather inputs from receptionist around the world to provide them an effective software-hardware combination. - no one around the world really targets receptionist tools market (is it a market ?) : some companies sell headphones or hardphones but receptionnist account for such a tiny part of sales that these companies cannot really hear receptionists demands and design specific products. - even if someone ever decide to focus on this, it would be difficult for someone to convince companies to improve receptionist tools once receptionist are trained and used. Maybe, a standard PC+headphone + a couple of software would be the best way to go ? Even on that, obstacles remain such as : - how do you monitor legacy PBX users along Asterisk users ? - how do you monitor a distant Asterisk server whitout any Data link between both locations ? Regards >From a sales perspective, one needs a system that is capable of many different configurations. a small business wants shared line appearances, since that has been proven to work for them. Our of an office, imagine a retail store, where everyone on the floor needs to be able to answer, redirect a call, not know if another station is busy or not, be able to do an all call page, or an off hook voice announce, to name a few features. Square hybrid works well here. In a larger establishment that has a receptionist who is used to a DSS/BLF, before the sale there is a good chance he/she will be consulted, and reject anything that requires a great deal of change. Soft BLF, as someone else pointed out, can be a real problem when the desktop is busy, has crashed, or the BLF window closed accidently Trying to jam someone into an IP system WILL meet with resistance. There are too many good systems out there that long ago overcame these problems, and many of them are NOT that expensive. Use the correct tool for the job. There are many places where Asterisk works, and many where it is a square peg in a round hole JMO John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
Dovid Bender wrote: snip Can Asterisk serve as an access server/gateway to the internet? /snip I have the same question. If I had a PRI coming in to asterisk can I have users dial in and have asterisk work as a gateway to the internet ? Dovid Very interesting question. if this feature is absent, is it possible to add a module for doing such thing ? And how hard will it be to implement it. Tofik Suleymanov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI issues
Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also notice that it is saying I don't have a D channel defined. I am not sure why it is saying that either. Below are my zapata.conf files. If anyone has any suggestions/ideas it would be greatly appreciated. Thanks, Kevin /etc/asterisk/zapata.conf switchtype=national defaultzone=us context=default signalling=pri_cpe group=1 channel = 1-23 dchannel=24 callerid=asreceived /etc/zapata.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Exception on 19, channel 2 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Exception on 18, channel 1 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Got event Alarm(4) on channel 1 (index 0) Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 Mar 29 17:08:18 WARNING[24038] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 3: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 3 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Got event Alarm(4) on channel 2 (index 0) Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 4: Red Alarm Mar 29 17:08:18 WARNING[15151] chan_zap.c: Detected alarm on channel 2: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 5: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 5 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 6: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 6 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 7: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 7 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 8: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 8 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 9: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 9 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 10: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 10 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 11: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 11 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 12: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 12 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 13: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 13 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 14: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 14 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 15: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 15 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 16: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 16 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 17: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to
RE: [Asterisk-Users] Quintum Tenor DX4060
I have four tenorAX boxes and there were way too many options that I would never use. Quintum has great support so use them. Only really tricky parts on the AX box was the dialplan section needed to be blank except min and max and the unit ships with g729 enabled which I changed to ulaw. Different boxes but same ideas I think. Thanks, Steve Totaro -Original Message- From: Stephen Arulraj [mailto:[EMAIL PROTECTED] Sent: Fri 3/31/2006 5:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] Quintum Tenor DX4060 Has anyone linked the Asterisk to the Quintum Tenor DX4060? If yes I would appreciate any valuable information to do this in anyway. Cheers Stephen winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiple auto attendants
Assuming your definition of 'auto attendant' is the same as my own, then you betcha! If I were building such a beast, I would use a different context for each tenant that wanted a customized IVR and a public/generic one for everyone else. You could use DID to route the incoming calls to the proper context. I would suggest that the public one run into a tenant-wide directory with voicemail. Or maybe I wouldn't. That doesn't seem very customer friendly. You will want to get all the requirements of this project lined out before you begin, as there are a lot of options here. You could conceivably waste a lot of time... Still, Asterisk can easily handle this for you. Good luck, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paolo Supino Sent: Thursday, March 30, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] multiple auto attendants Hi I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? TIA Paolo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Marketing Materials
Very true, but I want to 'sell' him on the idea, not drive him screaming to Cisco... Not my cup of tea, I'm afraid. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Friday, March 31, 2006 6:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Marketing Materials We created out own. You should do the same. --- Bob McDowell [EMAIL PROTECTED] wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX: Auto-congesting call due to slow response
It's been a while, but I didn't think those two terms were necessarily exclusive. Checkpoint firewalls can provide NAT, can they not? Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Friday, March 31, 2006 7:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX: Auto-congesting call due to slow response -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek I have same problem, do you have asterisk box behind nat? No, they are not behind NAT, peraphs there is a Checkpoint firewall. DV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Hylafax, on the same box
I agree, this does work well. My 'fax' extension is right off of the docs: - [faxin] exten = fax,1,UserEvent(Incoming Fax...) exten = fax,n,Dial(IAX2/ttyIAX) exten = fax,n,Dial(IAX2/ttyIAX2) exten = fax,n,Dial(IAX2/ttyIAX3) exten = fax,n,Dial(IAX2/ttyIAX4) exten = fax,n,Busy exten = fax,n,Hangup - Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Friday, March 31, 2006 7:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Hylafax, on the same box That's not entirely correct :) Fax and voice on the same DID is not possible when using a second application like hylafax. Because how should the two applications decide which one accepts the call? With the help of iaxmodem (which works really well) its easily done! Just detect the incoming call is fax and the route it to iaxmodem on fax extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI issues
Post your 'cat /proc/interrupts' for us. Kevin Smith wrote: Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also notice that it is saying I don't have a D channel defined. I am not sure why it is saying that either. Below are my zapata.conf files. If anyone has any suggestions/ideas it would be greatly appreciated. Thanks, Kevin /etc/asterisk/zapata.conf switchtype=national defaultzone=us context=default signalling=pri_cpe group=1 channel = 1-23 dchannel=24 callerid=asreceived /etc/zapata.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Exception on 19, channel 2 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Exception on 18, channel 1 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Got event Alarm(4) on channel 1 (index 0) Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 Mar 29 17:08:18 WARNING[24038] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 3: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 3 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Got event Alarm(4) on channel 2 (index 0) Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 4: Red Alarm Mar 29 17:08:18 WARNING[15151] chan_zap.c: Detected alarm on channel 2: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 5: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 5 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 6: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 6 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 7: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 7 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 8: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 8 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 9: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 9 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 10: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 10 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 11: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 11 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 12: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 12 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 13: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 13 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 14: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 14 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 15: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 15 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 16: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 16 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 17: Red
Re: [Asterisk-Users] PRI issues
Kevin Smith wrote: Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 5: Red Alarm Kevin, How exactly and to what are you hooking this up to? I've seen the above error when trying to hookup a Tellabs to our TE110P and didn't have the signaling correct. --span=1,0,0,esf,b8zs Also, this line says to get your timing from the Asterisk machine, and not the provider (If there is one) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Building Asterisk embedded device
Not a lot to go on sam. What do you want to do? If you just want to play or have very minimal requirements then get a soekris NET4801 board, CF and install Astlinux. http://www.soekris.com/net4801.htm -Original Message- From: sam [mailto:[EMAIL PROTECTED] Sent: Friday, March 31, 2006 6:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Building Asterisk embedded device Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Thanks A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple auto attendants
We're assuming you will use a T1 (or E1) for your PSTN interface. If you're using POTS lines then there will be no information about which number was called--you'll need a separate POTS line(s) for each tenant. We have multiple tenants on our hosted PBX without problem. I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX: Auto-congesting call due to slow response
Hi Domenico - I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to 'qualify=yes'... Yes. That's correct. That message is what is given when when the IAX peer fails to qualify. For whatever reason, one box is not able to reach the other for a short time. Qualify=yes will give you a default value in milliseconds (something like 500). If the peer doesn't respond within that time, you get the error message. It may just be temporary network hiccups, or one box may be very slow to respond due to high traffic. You can try increasing the qualify value like: qualify=1000. Otherwise, if you don't have any backup routes planned in your dialplan, you can just remove the qualify statement. Then it will just try the IAX link no matter what. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Confused on Agents and Queues
Hi, I'm confused with agents and queues in Asterisk. If I use AddQueueMember() then show queues shows the agents that I have logged into the queue... however the agent ID has to be the extension the agent is sitting at ... kinda useless for stats tracking. If I use AgentCallbackLogin() then show queues shows no agents logged in, but it works and show agents shows the agent logged in. How can I have my agents log in with a unique ID for *THEM* and have the calls ring to whatever extension they are at? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Referral - Cleanup on Aisle 7
Just got a call from a company in Warren, MI . They recently had an Asterisk system put in by a vendor, and are having issues which need analysis and correction. They have a tremendous sense of urgency. They have about (40) users, and need DIDs assigned to extensions and are having some echo issues at the site. If anyone is in the Warren, MI area, and is interested in some cavalry work, shoot me an email. Thanks, Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
Craig, Please correct the date on your machine. Your emails stick to the top of the list because they have a date of 6/30/2006. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
seems that if you get that log you didn't use jitetr buffer at all. In my opinion the latest jitter 1.2-branch is not working, the last working seems 1.2.1 patched. Hope Zoa could lead us to fix it. Regards Rosario - Original Message - From: Adam Moffett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 17, 2006 11:10 AM Subject: Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5 jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2 for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer I installed the jitterbuffer-1.2 branch and I have a few questions. First and foremost I'm getting hundreds of lines like this in my log file: Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 The console shows something similar: Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:10 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 My log file is going to be very big today. What could be responsible for frames (every frame?) having invalid timing info? Second I'm not sure if it's actually doing anything. For testing, I tried setting the max size to 2000ms and implementation to fixed.if I'm reading the comments in the sample config correctly that should create a 2000ms fixed jitter buffer, which in turn should mean a 2 second delay in audio, but I wasn't hearing any delay at all. Is this not a valid way to test whether the jitter buffer is doing something? ThirdI'm interested in a way to create some jitter ;) I was thinking I might take an ethernet hub and try to saturate it with several simultaneous large file transfers or something like that. Another possibility might be an 802.11 wireless connection at a fairly long range. If anyone knows of a more convenient way for me to create a jittery connection I'd be very interested. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming triggers seperate outbound
Hey, I would like in the course of dial plan logic, to trigger a separate outbound call. If that outbound call is answered, and if that certain key response is detected then it will bridge the incoming call to the newly dialed outbound call. What I want to accomplish is that when a caller dials in, they can enter enter an extension that will call out to a callee's cell phone. When the callee answers their cell they have to dial 111 or some other combo to accept the call. when this is done only then will the two calls be connected. Thanks Miles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXY codec support and questions..
I have been evaluating the Iaxy and Asterisk for the company I currently
work for, and am rather impressed with them both. Once configured, the
Iaxy is a solid device--it's pretty much an appliance at that point
(plug it in, turn it on, and leave it alone).
My only gripe is the initial configuration, although even that isn't too
terribly bad. You must download and unpack a C program, then edit a
config file that the C program pushes to the Iaxy. If you want to change
settings on the Iaxy, you must reset it (press the reset button on the
back, hold it for ten seconds, unplug the RJ-11, RJ-45 and power cables
from the Iaxy while holding the reset button, then replace the RJ-11,
RJ-45 and power cables) before you can push the new config to the Iaxy.
The Iaxy I am using is inside the same RFC-1918 network as my Asterisk
server (inside the NAT, that is), but I have used an IAX softphone from
inside another RFC-1918 network to place calls through my Asterisk
server, and that worked just fine, so I suspect the Iaxy would work just
fine through double NAT as well.
--Mike Wallette
Date: Fri, 31 Mar 2006 15:57:28 +0100
From: WipeOut [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAXY codec support and questions..
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi..
I have to setup an extension in a remote location that will use a
cordless analog telephone.. I am looking at the IAXY to do this for
me..Basically the data path will be as follows...
[Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset
Since there are two NAT boxes in the path I know SIP won't work.. I also
don't want to move the Asterisk box to the internet side of the NAT box,
not only from the security perspective but also the potential issues
with the already configured SIP phones that connect to it locally..
So as far as getting over the NAT problem the IAXY seems the way to go..
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
Thanks for any suggestions or input on this setup.. Also any reviews on
the IAXY are welcome..
--
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[Asterisk-Users] Transcoding on asterisk
Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I dont have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up From [cisco (g729)] [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] - [pstn] And vice versa. I will be glad if someone can throw more light on this for me. Goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Monday, March 27, 2006 3:19 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] g723 or g729 on ss7 link Hi! Is it possible to set codecs on ss7 link? No. E1s channels (which chan_ss7 uses as voice channels) can only use G711 alaw. Or receiving call with g723 or g729 and forward the call to pstn via the ss7 link. You can do transcoding on the asterisk machine that you use as SS7 Gateway. This means you get the calls delivered via SIP with G723 and send them out to SS7 with G711. Note that you need more processingpower for this then if you do no transcoding. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-14 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
