Re: [Asterisk-Users] Session timer
Thank you for a reply, Mr. bkw. Although Session Timer is the Internet draft now, I think that a possibility of being set to RFC is high. Is there any schedule whose Asterisk supports Session Timer? Nope, But if you check the latest sip.conf.sample ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be rtptimeout) This should solve your problems. It is the Session Timer option function itself which I expect. Therefore, now, it did not solve. bkw - Original Message - From: Ichiro Nakata [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 12:05 AM Subject: [Asterisk-Users] Session timer There is one question about re-Invite. Is it possible to carry out operation corresponding to draft-ietf-sip-session-timer -14? Ichiro Nakata [EMAIL PROTECTED] Ichiro Nakata [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] strange problem with oh323 loaded!
Hi, Anthony, can you try issuing stop now on safe_asterisk and see if it works please? I am used to using safe_asterisk and with this new version and when I tried issuing stop now, it did not do it. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anthony Law Sent: Wednesday, June 30, 2004 4:17 PM To: Mailing List Asterisk Subject: [Asterisk-Users] strange problem with oh323 loaded! Hi, I am using asterisk CVS 2004-06-16 with oh323-0.6.3a I have a strange problem if I start asterisk with oh323 loaded /usr/sbin/asterisk -vc once I am in the console and issue restart now or reload asterisk hangs and it not stoping or restarting at all, below is the console logging when it happens, as you can see it stucks on Destroying any remaining musiconhold processes [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.6.3) == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Asterisk Ready. *CLI restart now Beginning asterisk restart Executing last minute cleanups == Cleaning up OpenH323 channel driver. == Unregistered channel type 'OH323' == Destroying any remaining musiconhold processes If I do not load oh323 the above will not happen. Does anyone knows how to why or how to fix? Even if I use safe_asterisk it acts the same. Is this a problem with oh323 or asterisk itself? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Delivery from China
That would be a great alternative. For what it's worth, the phone is based on a PA1688 single-chip VOIP terminal, which in turn contains a 50MHz 8051-compatible and a ADSP2181 DSP running at 33MHz. Okay, open sourcers, that does not include Linux. Even uLinux (that runs on CPUs without a MMU) should be far to fat for this environment. Hey, that thing has even still Banks to access memory, very much like the Lotus EMS that we once used years ago on 8086 and 80186. Or in the Language Card for the Apple II ... For what it's worth, I was able to determine that they're using VC6 and KeilC51 (?) to cross-compile. Keil is a company that develops and sells cross-compilers for a host of embedded type CPUs. The compiler usually runs on Windows and generates binary files that you either flash into Flash chips, EEPROM or via JTAG. It's well known in the commercial community. The KeilC51 costs here 1600 Euro, and that's just the CA51 Compiler+Assembler. No debugger. I think that the No Linux and Windows words in my statemement above greatly reduces the chance that people really will jump onto this opensource bandwagon. The price tag as well (althought me might be able to create a 8051 cross compilation environment on Linux). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 01/07/2004, at 5:21 AM, programmer_ted wrote: By the way, how do you like your BT101? My friend (mentioned in the bugfix email) ordered one to use with FWD, and for a cheap SIP phone, it seems to work very well. Looks pretty good, too (black). We bought 8 BT101, works very well. A pity they're only using 10mbit/s interface. Web interface is rather primitive but it's functional. It's lacking some feature like if you transfer a call, you can't talk to the destination first then transfer. You have to transfer if blindly. Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA47lAXeDVKqIr3GURAlkhAJ4p/IZaIo34pffsEbuKDE9zrpybywCeJySH KdJ2moBkFAjUx3xnMwCTmJU= =vtcJ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
IS VONAGE LISTENING? RE: [Asterisk-Users] Vonage and Asterisk integration
On Tue, 29 Jun 2004, Jay Milk wrote: Like I said, they just seem to be lazy and/or badly organized. If they can do LNP, why can't they change a hardline into a softphone, break one number out onto a different ATA, etc? I basically laid it out for them, saying If you can't move my 2nd line from this ATA to a new ATA, then I'll need to cancel that line... I no longer have that line. Not being able to something this simple cost them over $500/year from me... I wonder how many other Vonage users will drop them because of such things. We were considering Vonage - but if this is the case - we have no interest. Is vinage listening? regards joe -Original Message- From: Steve Kalcevich [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Vonage and Asterisk integration Jay Milk wrote: I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. I never worked with vonage, is there tech support that bad? -- Regards, Steve Kalcevich, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 T100P and a Digital PBX
I need help configuring a 2 Digium Wildcards with Asterisks and a Digital PBX. MY Goal: CO === ASTERISK === Digital PBX Current PBX already works with PRI ISDN My preliminary attempts when I run ztcfg I get error below ZT_SPANCONFIG failed on span 1: Invalid argument (22) /etc/zaptel.conf #Signaling for 1 X100P Wildcards. fxsks = 1 #Signaling for T100P Wildcards. span=1,1,0,esf,b8zs bchan=2-24 dchan=25 span=3,0,0,esf,b8zs fxoks=26-49 Anyone knows what I am doing wrong. I am new to T100P but comfortable with the X100P setups ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
I sent this bugfix to the asterisk-dev mailing list, [...]. Nobody seemed to notice it there, so I thought I'd post here, Please file a bug report at http://bugs.digium.com and attach your bug fix. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
It does not work with the binary only AVM Fritz card driver. And I did not get P2P+DID working even with active AVM card. The chan_capi driver kept spindle a loop when I started Asterisk. Now I'm at zaphfc, that works the best. So, to make things simple: If you want P2P+DID (Anlagenanschluss), you won't go for AVM cards to save nerves (and money, the active cards are not cheap). Go for zaphfc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip to isdn-capi call problem
Klaus-Peter Junghanns wrote: Hi Tomaz, make sure you disable the G723.1 codec in your SIP device, asterisk does not support G723.1. Use G711 (alaw, ulaw)! best regards Klaus yes ,this was a problem . thank you. tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Delivery from China
Can we see a picture of this thing? One pic is worth 1000 words etc... Rgds Tim Jay Milk wrote: That would be a great alternative. For what it's worth, the phone is based on a PA1688 single-chip VOIP terminal, which in turn contains a 50MHz 8051-compatible and a ADSP2181 DSP running at 33MHz. The Sound interface is AC97 compatible, the network interface is NE2000 compatible (RTL8019 chip), running only 10mbps. For what it's worth, I was able to determine that they're using VC6 and KeilC51 (?) to cross-compile. -Original Message- From: James H. Thompson [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 11:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Special Delivery from China Another approach would be to sell the hardware without firmware and start and opensource project to build firmware for it. It would seem like this could be a good niche for a small manufacturing company. Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP-Asterisk-GnuGK-Cisco 5300
Ganbaa a écrit : Hi Thank you for your response and advice. I did between h323 EP, gnugk and cisco as5300. Now I'm trying to test Asterisk as translator (SIP-H323). So I need sample config for asterisk and gnugk. Could you give me advice? There is a h323.conf sample file in sources Ganbaa - Original Message - From: administrator tootai [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 5:43 PM Subject: Re: [Asterisk-Users] SIP-Asterisk-GnuGK-Cisco 5300 Ganbaa a écrit : Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Yes. Setup your Cisco as EP in gnuGk, and use the h323 channel from * to redirect call to GnuGK. -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash?
On Wed, Jun 30, 2004 at 05:05:26PM -0400, Brian Wilkins wrote: Date: Wed, 30 Jun 2004 17:05:26 -0400 From: Brian Wilkins [EMAIL PROTECTED] Organization: HCC User-Agent: KMail/1.6.2 To: Asterisk-users [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash? Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we had the Asterisk server on an exact same router but different network before and it did not cause a crash. We have gone through two different Cisco 7200 series routers and both exhibited the same problems. Any clues? Thanks - I think you should open a TAC case on cisco or contact your cisco representative. IMHO it's a serious problem, if you can crash your cisco just by starting asterisk. BTW, have you saved cisco's configuration in nvram after configuring it. How cisco is configured? is it just ip gateway or you are using it as Voice gateway? in second case what hardware: BRI/PRI? what protocol h323/mgcp? -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
Holger Schurig wrote: It does not work with the binary only AVM Fritz card driver. And I did not get P2P+DID working even with active AVM card. The chan_capi driver kept spindle a loop when I started Asterisk. Now I'm at zaphfc, that works the best. So, to make things simple: If you want P2P+DID (Anlagenanschluss), you won't go for AVM cards to save nerves (and money, the active cards are not cheap). Go for zaphfc. Just one thing ... for zaphfc .. what version of kernel is best option 2.4.26(28) or 2.6.7 ... for bri-stuff ? what must be compiled in kernel or what modules must be selected (i mean isdn stuff) ? if any .. Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directing to different Voicemailboxes by Callermsn?
Hi List! I am not sure, that my questin reached the list, so here I go again: I would like to have: - a Voicemailbox with 1 MSN - Callers calling this Voicemailbox are directed to differen Voicmailbox Extensions, depending on their (_THEIR_) MSN. Is this possible in any way? I have tried differentiating this by incomingmsn in capi.conf, but that didn't work. Any ideas would be appreciated, thanks! Cu Henning ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID
Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send "other numbers" as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 30 June 2004 18:57To: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve
RE: [Asterisk-Users] Providing Telewest in the UK with per extens ion outbound callerID
Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send other numbers as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman Sent: 30 June 2004 18:57 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID
Hi Steve, Telewest should already allow the CLI transmission of your DDI range, without further datafill changes. If it doesn't work you should check that you are sending the appropriate number of digits. Try sending: -3 digit CLI -the whole number (minus the leading zero) If the comments above don't help please post a trace of an outgoing call and detail the number, if any, that is presented to theCalled Party. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 01 July 2004 09:57To: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message-From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send "other numbers" as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 30 June 2004 18:57To: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] Providing Telewest in the UK with per extens ion outbound callerID
When the original PBX was installed we asked them to override the CLI and provide a single number as the PBX couldn't provide the DDI number, now the contact at Telewest believes it's somewhere between illegal and impossible to provide DDI numbers to the outside world. -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 10:13 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, Telewest should already allow the CLI transmission of your DDI range, without further datafill changes. If it doesn't work you should check that you are sending the appropriate number of digits. Try sending: -3 digit CLI -the whole number (minus the leading zero) If the comments above don't help please post a trace of an outgoing call and detail the number, if any, that is presented to theCalled Party. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman Sent: 01 July 2004 09:57 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send other numbers as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman Sent: 30 June 2004 18:57 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID
SHthe contact at Telewest believes it's somewhere between SHillegal and impossible to provide DDI numbers to the outside world. Complete nonsense, ask to speak with someone from the Datafill Department. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 01 July 2004 10:16To: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID When the original PBX was installed we asked them to override the CLI and provide a single number as the PBX couldn't provide the DDI number, now the contact at Telewest believes it's somewhere between illegal and impossible to provide DDI numbers to the outside world. -Original Message-From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 10:13To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, Telewest should already allow the CLI transmission of your DDI range, without further datafill changes. If it doesn't work you should check that you are sending the appropriate number of digits. Try sending: -3 digit CLI -the whole number (minus the leading zero) If the comments above don't help please post a trace of an outgoing call and detail the number, if any, that is presented to theCalled Party. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 01 July 2004 09:57To: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message-From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send "other numbers" as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 30 June 2004 18:57To: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error
[Asterisk-Users] Simple gateway SIP -- ISDN
Hi *, I have a very simple setup, since this is my first test with asterisk: I have configured an Asterisk server and a kphone client (SIP) to talk to each other. Right now, the SIP user gets authenticated by asterisk without problems. My goal is to redirect the call to a given ISDN telephone number. Here are the parameters that I want to use for my setup: - SIP user: test_sip_user - destination ISDN telephone number: 123456 - asterisk server: 192.168.1.100 ISDN interface extension associated to the ISDN number: 100 - kphone client: 192.168.1.200 sound system (ALSA with OSS emulation, working) The first problem that I have is that, even though kphone and asterisk are able to authenticate the user, I am not sure that sound gets transmitted. This is the first thing that I would like to achieve: to verify that sound is flowing between kphone and asterisk. The easiest thing would be to get a dial tone in the kphone client, but I fear that this is not possible, since SIP initiates a session with all needed parameters, and does not need/accept a dial tone. Please, correct me (and tell me how to do it :) ) if I am wrong on this one. The next method to verify the flow of sound, easy enough for me to try, would be to set up a single mailbox, with a greeting message and the possibility to record speech on the mailbox. This should allow me to verify the flow of sound if both directions. Could you provide any hints on how to do this? Just a very simple setup is needed. Once I have verified that sound is flowing, I would like to make the call into the ISDN network. I have some questions: 1) Is it actually possible to implement this scenario? I have understood that asterisk can work as a gateway between SIP and ISDN (and between other networks, too). Is this correct? 2) I am not able to figure out what extension to use for the SIP user. The kphone sends the following request to asterisk: sip:[EMAIL PROTECTED]:5060 I do not know how to use this in an extension specification in order to get asterisk to dial the desired number (123456) via the ISDN interface. I have tried to setup extension 100 to playback a sound file, like this: exten = 100,1,Wait(1) exten = 100,2,Playback(demo-congrats) exten = 100,3,Hangup but kphone complains that the session can not be established. What extension specification should I use to match the SIP call? And I have an aside question: kphone can (apparently) also be used for video-conferences. Is this in any way supported by asterisk? My impression is that asterisk only provides voice services. Thanks for your help, Daniel Gonzalez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID
Hi Steve, Try Telewest Provisioning Dept. on: 01483 582 966 HTH Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 01 July 2004 10:16To: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID When the original PBX was installed we asked them to override the CLI and provide a single number as the PBX couldn't provide the DDI number, now the contact at Telewest believes it's somewhere between illegal and impossible to provide DDI numbers to the outside world. -Original Message-From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 10:13To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, Telewest should already allow the CLI transmission of your DDI range, without further datafill changes. If it doesn't work you should check that you are sending the appropriate number of digits. Try sending: -3 digit CLI -the whole number (minus the leading zero) If the comments above don't help please post a trace of an outgoing call and detail the number, if any, that is presented to theCalled Party. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 01 July 2004 09:57To: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message-From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send "other numbers" as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve HanselmanSent: 30 June 2004 18:57To: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you
RE: [Asterisk-Users] Re:Latest Echo changes
Just received it today - ultra fast shipping from digium. Will let people know the results of echo when I switch it tonight. -Original Message- From: Chris Bond [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 4:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re:Latest Echo changes Just spoke to someone at telappliant and there not willing to sell the cards in the uk yet as there not ratified to the UK standard. I've just spoke to someone at digium direct and there forfilling backorders at the moment. I've just placed an order at http://store.yahoo.com/asteriskpbx/newitd1pofxo.html. The guy recokens I they should start shipping at the end of the week. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Directing to different Voicemailboxes by Callermsn?
Hi List
Am Do, den 01.07.2004 schrieb Henning Vogt um 10:34:
I would like to have:
- a Voicemailbox with 1 MSN
- Callers calling this Voicemailbox are directed to differen Voicmailbox
Extensions, depending on their (_THEIR_) MSN.
You should be able to do this with the GotoIF Statement in
extensions.conf
e.g.
exten = 1234,1,GotoIf($[${CALLERIDNUM} = 5678]?3:2)
exten = 1234,2,GotoIf($[${CALLERIDNUM} = 9011]?4:5)
exten = 1234,3,VoiceMail(u1)
exten = 1234,4,VoiceMail(u2)
exten = 1234,5,VoiceMail(u3)
If the calling number is 5678, the call is forwarded to mailbox 1, if
the number is 9011, the call is forward to mailbox 2. In all other
cases, the call is sent to mailbox 3.
Hope I gave you a vague direction. More about GotoIf can be found in the
wiki...
Best regards
Kai
--
Kai Militzer WESTEND GmbH | Internet-Business-Provider
Technik CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10 Tel 0241/701333-11
[EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879
___
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[Asterisk-Users] Help with Welltech 2FXO gateway, GS BT100 and Asterisk
Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of problem and what should be the solution about this. I already upgrade the firmware of my Welltech 2FXO to version 103 but problem still the same. Is this an Asterisk related issue or maybe on my config? I hope anyone can open up the solution or should I post my config for further info about my settings. thanks in advance. regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Welltech 2FXO gateway, GS BT100 and Asterisk (FIXED)
Hi all, Just to follow up on my post. I just fixed the problem by removing the dtmfmode entry on my sip.conf, voice works well now. regards - Original Message - From: Glynn Condez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 6:28 PM Subject: [Asterisk-Users] Help with Welltech 2FXO gateway, GS BT100 and Asterisk Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of problem and what should be the solution about this. I already upgrade the firmware of my Welltech 2FXO to version 103 but problem still the same. Is this an Asterisk related issue or maybe on my config? I hope anyone can open up the solution or should I post my config for further info about my settings. thanks in advance. regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
Dear Ted
i have notice the same problem had you reported from monday, i have try to
update to today CVS HEAD but nothing still buggy so i roolback to Stable V1.
Where i can find the pacth?
Thanks in advance
Dimitri
On Wednesday 30 June 2004 08:39 pm, programmer_ted wrote:
Hiya,
I sent this bugfix to the asterisk-dev mailing list, and modified it as I
noticed side effects, but now it appears to be finished. Nobody seemed to
notice it there, so I thought I'd post here, as it seems to be something
that will be needed as people update to the latest CVS version. So...read
on :)
Ted
[EMAIL PROTECTED]
P.S. Read to the very end. The original bugfix has an annoying side
effect.
Hi,
My friend and I were getting a warning when calling his Sipura from a
PSTN line (connecting to Asterisk through BroadVoice), that said:
Asked to transmit frame type 64, while native formats is 4 (read/write
= 4/4)
and was followed by a hangup (type 64 is 16-bit Signed Linear PCM,
type 4 is G711u). I found that many people have had similar issues,
but these were never resolved. So, I figured that because Asterisk is
open-source, I'd dive into the code and try to fix the bug.
After a couple of hours of familiarizing myself with the Asterisk code
and tracing the problem, I found that for some reason the tone
generator, which uses 16-bit Signed Linear PCM, was still being
allocated and playtones_generator (indications.c) was still getting
called, regardless that the Sipura doesn't take SLINEAR data (in my
case, it accepts G711u). So, I ended up adding an if conditional to
the beginning of the playtones_alloc function (indications.c) to check
if SLINEAR was supported by the channel, and if not, return 0 (which,
when received by the ast_activate_generator function (channel.c),
causes the channel generatordata to remain empty, effectively stopping
the SLINEAR data from being sent in the most nonintrusive way
possible).
NOTICE: this bugfix will work for similar issues involving format 64
(16-bit Signed Linear PCM) being sent even if channel capabilities
don't allow it, if the generator is involved - it's not limited to my
situation (dialing the Sipura from Asterisk).
This patch should be applied to indications.c under the main asterisk
source directory (usually /usr/src/asterisk):
68a69
if (!(chan-nativeformats AST_FORMAT_SLINEAR)) return 0;
Oh, and finally, here's a shameless plug to a good friend's website
(which includes a VOIP forum!): http://outcast.ws
Comments? Questions? :)
Just a quick update. I was looking things over again and it appears
this fix also disables the generator when I'm calling in on PSTN over
BroadVoice (when dialing the Sipura), not just disabling it for the
Sipura. This basically disables the dialing sound while waiting for
the Sipura to pick up. I have an idea that I should have used
chan-capabilities rather than chan-nativeformats, but it's too late
to check at the moment. I'll try it out first thing tomorrow and
update you guys, but for now, that's one drawback of using this fix.
I thought it over a little bit more and the optimum solution would be to
just translate the SLINEAR data to a format that is recognized by
whoever is receiving the data, thus eliminating all drawbacks. I'm
going to try using capabilities rather than nativeformats as a quick
workaround (after debugging to see if it'll work), and then work on
adding the translating code to sip_write. Actually, thinking about it
again, it'd probably be best to just translate at the
playtones_generator function. I'll keep you guys updated.
...snipped non-relevant signature info etc...
Learning as I go. It appears I don't have access to the capabilities
value from the ast_channel structure. I'm just gonna go ahead and have
the SLINEAR data translate to the channel's writeformat.
Ok, as I thought, PSTN over BroadVoice does not understand SLINEAR
natively, which is why the dialing sound was also disabled when I dialed
the Sipura. I added some code to playtones_alloc (indications.c) so that
the write format is only set to SLINEAR if it's supported, and added some
code to playtones_generator to translate from SLINEAR to the channel's
writeformat if SLINEAR isn't supported natively by the channel. Of
course, I also had to include the translate.h header.
Conclusion: playtones_generator now works regardless of SLINEAR support by
the channel, as long as a translator path can be found from SLINEAR to the
channel's writeformat. If SLINEAR is supported, no translation takes
place. This should fix some bugs where format 64 is being sent regardless
of codec allow settings in the configuration files.
Apply this patch to indications.c:
28a29
#include asterisk/translate.h /* Needed for bugfix */
75c76
if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
---
if ((chan-nativeformats
Re:[Asterisk-Users] QoS in Cisco
What IOS version has a fix for this bug and what IOS should work in QoS in ethernet. regards I remember seeing a notice about a fix about a month ago, don't remember any specifics. The actual bug was a weird one and required simultaneous use of QoS output service policies, PBR, and multicast PIM-DM to happen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto: Use setgroup, checkgroup to check incoming and outgoing client limits
Well that works.. But lets say I wont to be able to
control incoming and outgoing limits on all channels. I have3 phones
registered and phone 1 calls phone 2. With the example below phone 1 cannot make
anymore calls.. But phone 2 can (even though stíll talking with phone
1)
Phone 2 can also still receive anothercall
from phone 3.
exten = s,1,SetGroup(SIP/${CALLERIDNUM}) ;Check
to see if outgoinglimit of caller has been reachedexten =
s,2,CheckGroup(1)exten = s,3,Dial(SIP/${ARG1}, 30, tr)exten =
s,4,Goto(s-${DIALSTATUS}, 1)exten = s,103,Hangupexten =
s-NOANSWER,1,Voicemail(u${ARG2})exten = s-NOANSWER,2,Hangupexten
= s-CHANUNAVAIL,1,Voicemail(u${ARG2})exten =
s-CHANUNAVAIL,2,Hangupexten = s-BUSY,1,Voicemail(b${ARG2})exten
= s-BUSY,2,Hangup
Help PLEASE!
Claus
- Original Message -
From:
Jason Williams
To: [EMAIL PROTECTED]
Sent: Friday, June 25, 2004 1:52 PM
Subject: Re: [Asterisk-Users] Howto: Use
setgroup, checkgroup to check incoming and outgoing client limits
At 13:00 25/06/2004 +0200, you wrote:
Hi there,I was wondering
how I can use setgroup and checkgroup for perfom incomingand outgoing
limitation checks.I've have some users that doesn't what to be able to
recieve more than 1call at a time, and I also want to limit a users
outgoing call abilities.Any help would be greatly
appreciated.exten =
999,1,SetGroup(moh);Set
Current Group to moh
exten =
999,2,CheckGroup(1);Check
moh does not have more than 1
exten =
999,3,Answer;Answer
the call
exten =
999,4,MusicOnHold(default);Play
default Music on hold
exten =
999,103,Busy;Play
busy if 1 person is already listening
This will allow only one call to use
the resource music on hold.Jason
---Outgoing mail
is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.708
/ Virus Database: 464 - Release Date:
19-06-2004
RE: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash?
As far as loosing the configuration...the only reason I could see that happening is if you either are doing one of the two... not saving the configuration...or you have the configuration register set to something like 0x2142. look on show version for the configuration register. it should be 0x2102. And again, i would look for tracebacks...it could either be a memory issue or a bug in the IOS. But you will know if you get console access to the router as u bring up the asterisk... As a side note, Cisco has had various IOS bugs over the years where certain parameters were never written to nvram. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P in Japan (eh?)
I'm planning to buy a T100P for a project in the company where I work for but my concern is about the japanese ANI. Can I get somehow japanese(NTT) ANI working with T100P ? Feasible? Impossible ? Thanks, Isamar Maia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Patch for call queues?
At 00:35:41, CW_ASN wrote: Please try CVS, AFAIK patch 214 doesn't included in stable branch. But I need to apply some other patches too that isn't included in the CVS! How can I do that when I install * CVS? Best regards, Robin -- Robin Calmegård Siurua CEO/developer RoCaS - development support tel +46 8 505 556 80 fax +46 8 505 556 79 mobile +46 73 643 68 05 [EMAIL PROTECTED] www.rocas.se ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco phone and parked calls
Brian K. West [EMAIL PROTECTED] wrote: Two words... Valet Parking... You'll probably find that more than two words are required in this case, especially as app_valetparking.c is not in CVS. For instance, you could have told the original poster that the Valet Parking application can be found here: http://www.loligo.com/asterisk/misc/apps/ The app_valetparking.c says Copyright (C) 1999, Mark Spencer, so I imagine that there's some issue that's prevented him from putting the code into CVS. I haven't tried the application, so I don't know whether it works. I do know that there have been various mutex changes that don't appear to have been mirrored in that code. Perhaps it's not being maintained any longer. Perhaps there are plans to merge any new facilities into the existing parking application. I don't know. Unless the second word is off, two words are rarely enough to get a point across. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Mike, Contact me off list and let's see if we can isolate the issue. Can't tell from the words you've used what steps you've gone through to date. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP-Asterisk-GnuGK-Cisco 5300
On Wed, 2004-06-30 at 20:42, Ganbaa wrote: Hi Thank you for your response and advice. I did between h323 EP, gnugk and cisco as5300. Now I'm trying to test Asterisk as translator (SIP-H323). So I need sample config for asterisk and gnugk. Could you give me advice? Ganbaa - Original Message - From: administrator tootai [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 5:43 PM Subject: Re: [Asterisk-Users] SIP-Asterisk-GnuGK-Cisco 5300 Ganbaa a écrit : Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Yes. Setup your Cisco as EP in gnuGk, and use the h323 channel from * to redirect call to GnuGK. -- Daniel in extensions.conf [sip ext] exten=10,1,dial(oh323/number1@gnugkip) Where number1 is a number that your gnugk knows how to manage. That should do the trick. Make sure the context is one used by the sip phone. To test dial 10. Before doing this check the oh323.conf and make sure you uncomment the default values so it gets loaded. -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
All voip providers will use digital 4-wire interconnect to Asterisk or similar, so echo problems are much reduced, as there are only 'echo points' at the far end and your handset. Using an analogue card will always have its issues if you have significant propagation delay in the path anywhere. You will see much bigger echo problems where the line is mismatched to the analogue card. This will be in most places other than the US, as the analogue cards only seem to support a 600 ohm line impedance. If you want my recommendation, abandon analogue and get a basic rate ISDN line and a zaphfc card. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: 01 July 2004 14:21 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next? Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Mike, Contact me off list and let's see if we can isolate the issue. Can't tell from the words you've used what steps you've gone through to date. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] Patch for call queues?
It's included in CVS. I'm using it from there! Anyway, the patch is 214. Look http://bugs.digium.com/bug_view_page.php?bug_id=214 Regards, Gus At 00:35:41, CW_ASN wrote: Please try CVS, AFAIK patch 214 doesn't included in stable branch. But I need to apply some other patches too that isn't included in the CVS! How can I do that when I install * CVS? Best regards, Robin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registration failed for SIP
I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? How Can be fixed? message : Jul 1 16:18:29 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'damian sip:[EMAIL PROTECTED]' failed for '10.1.1.11' Asterisk and Sip phones are all in one network , no nat. Here is the Config in sip.conf [phone1010] type=friend host=10.1.1.11 auth=md5 nat=no reinvite=no canreinvite=no qualify=1000 dtmf=inband callerid=Damian Minkov 1010 username=damian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Docs
OK, this may seem to be an obvious question but where do I find
the reference docs? I'm getting this error message:
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know what the
parameters mean. I looking for a man page basically.
--
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[Asterisk-Users] voicemail notification?
Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a second email address (eg, pager), it works but the first address does not, like: 3002 = 3002,Rich,[EMAIL PROTECTED],[EMAIL PROTECTED] Played with the context to ensure that wasn't an issue. Faintly remember seeing something modified via cvs list, but can't seem to find anything addressing this one. Google doesn't provide any hints. Thoughts? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] execute a context from cron
I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. I figured that one way to do it would be to have cron run asterisk -rxsome command if there were some command that would tell asterisk to go to a specific context,extension,priority, but I cannot find that command. Does such a command exist? Or is there a better way to do this that I have overlooked? Thank you! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration failed for SIP
On Thu, 2004-07-01 at 16:24 +0300, Damian Minkov wrote: I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? Registration is for dynamic clients. How Can be fixed? Either stop the client registering or really make it dynamic. [phone1010] type=friend host=10.1.1.11 host=dynamic defaultip=10.1.1.11 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] execute a context from cron
I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job Regards -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Asterisk Docs
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know
what the parameters mean. I looking for a man page basically.
The problem isn't related to the t in the Dial() command, which enables call
transfer, but to a missing t (timeout) extension. More can be found here:
http://www.voip-info.org/wiki-Asterisk+standard+extensions
The voip-info.org site is a good reference if you're looking for something like a man
page for Asterisk.
-Manuel
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[Asterisk-Users] 1800 number with colo
Hi all Was wondering if anyone is aware of a colo provider who can terminate a 1800 phone line to my box in their colo. I just need one or may be two phone lines with the same 1800 number to go to my asterisk box. Thanks for any help Hariom Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!
Re: [Asterisk-Users] execute a context from cron
On Thu, 1 Jul 2004, Michael George waxed: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. I figured that one way to do it would be to have cron run asterisk -rxsome command if there were some command that would tell asterisk to go to a specific context,extension,priority, but I cannot find that command. Does such a command exist? Or is there a better way to do this that I have overlooked? Try looking at sample.call in the top asterisk source directory. Set up cron to create this file to connect to the specific extention and dump it into: /var/spool/asterisk/outgoing --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Docs
Neil Cherry [EMAIL PROTECTED] wrote:
OK, this may seem to be an obvious question but where do I find
the reference docs? I'm getting this error message:
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know what the
parameters mean. I looking for a man page basically.
It has nothing to do with the 't' in your Dial().
The Dial() command docs can be found here:
http://www.voip-info.org/wiki-Asterisk+cmd+dial
The predefined extension names list, including 't', can be found in
here:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
The 't' context is called when a timeout occurs. You could get rid
of the warning with the following:
exten = t,1,Hangup
That would simply hang up the line when a timeout is detected. You
could do anything you like in there, of course.
This page could be helpful too:
http://www.voip-info.org/wiki-Asterisk+cmd+ResponseTimeout
The WiKi is your friend.
--
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Re: [Asterisk-Users] execute a context from cron
On Thu, 2004-07-01 at 09:46, Michael George wrote: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. I figured that one way to do it would be to have cron run asterisk -rxsome command if there were some command that would tell asterisk to go to a specific context,extension,priority, but I cannot find that command. Does such a command exist? Or is there a better way to do this that I have overlooked? http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20openhours ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] execute a context from cron
Manuel Wenger [EMAIL PROTECTED] wrote: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job I prefer the 'include' method, personally, as explained here: http://www.voip-info.org/wiki-Asterisk+tips+openhours the choice is yours. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] execute a context from cron
On Thu, Jul 01, 2004 at 03:58:25PM +0200, Manuel Wenger wrote: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job It looks like this is just a conditional Goto and it will not spontaneously start a flow in a context. What I need is something that will, at a given time, act just like we picked up an internal extension and dialed a sequence of numbers. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1800 number with colo
Hariharan Gopalan [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Was wondering if anyone is aware of a colo provider who can terminate a 1800 phone line to my box in their colo. I just need one or may be two phone lines with the same 1800 number to go to my asterisk box. Someone may offer to set you up in their colo facility. If not then why do you need it at all? You could get a freephone number from a company such as NuFone (www.nufone.net) and have calls routed to your server located anywhere in the world. Do you Yahoo!? No. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1800 number with colo
I just started with coloco.com , so far so good. Rob - Original Message - From: Hariharan Gopalan To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 10:07 AM Subject: [Asterisk-Users] 1800 number with colo Hi all Was wondering if anyone is aware of a colo provider who can terminate a 1800 phone line to my box in their colo. I just need one or may be two phone lines with the same 1800 number to go to my asterisk box. Thanks for any help Hariom Do you Yahoo!?New and Improved Yahoo! Mail - 100MB free storage!
Re: [Asterisk-Users] Registration failed for SIP
But I've tried with these settings host=dynamic defaultip=10.1.1.11 But Again this notice. Is this possible - Not to mention the client IP , just host=dynamic Dave Cotton wrote: On Thu, 2004-07-01 at 16:24 +0300, Damian Minkov wrote: I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? Registration is for dynamic clients. How Can be fixed? Either stop the client registering or really make it dynamic. [phone1010] type=friend host=10.1.1.11 host=dynamic defaultip=10.1.1.11 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Blank faxes with RxFAX
Okay, so I finally found the problem that I was having with RxFAX receiving blank and mangled faxes. It turns out that it was caused by the timing source set wrong. I have a TE405P with span 1 running to a channel bank, a PRI (which the faxes were coming over) running into span 2, and an internet T1 running into span 3. I had span 3 set as the timing source. I changed timing to 0 on span 3 and set timing to 1 on span 2, rebooted and everything works perfectly. Hopefully, this will help some of the other people that were having the same problems. Patrick -- This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
On Thursday 01 July 2004 08:39, Robinson Tim-W10277 wrote: All voip providers will use digital 4-wire interconnect to Asterisk or similar, so echo problems are much reduced, as there are only 'echo points' at the far end and your handset. And on my PRI that is specifically where my echo is coming from... the far end. VOIP calls through nufone have no echo MOST PSTN calls through the PRI have no echo SOME PSTN calls (usually to local numbers NOT terminated at my local CO) have significant echo... I too have been unsuccessful in getting this zapped. My connection: Norstart MICS -- Adit600 --- T100P -- IAX2 -- TE405P -- Bell Canada PRI *1 = Xeon/2.4 with HT with T100P *2 = Xeon/2.4 with HT with TE405P *2 also does the NuFone IAX2 connection (it is always in the loop, as *1 is on a private network) Strange stuff, I am going to look at T1 echo cancellation hardware if I cant' get this solved. Tried: - echotraining=800 on *1 and *2 - echocancel=32,64,128 on both Eventually the MICS will have a digital connection to *1 instead of going through the Adit600 but we haven't got there yet :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
Hi, We have our own algorithm handling (dial plan) the calls and different events. When we receive an external call (from FXO), probably in consequence of our algorithm, some times the FXO port remains open and we could not establish the reason why the port is not closing. We were thinking a lot what might be the problem - for example we might forget to call the hang-up method somewhere in the script. Unfortunately we were not able to fix the problem. We came to the conclusion that the only way to establish where the mistake is, is to ask you for information about is there any log files, which could help us tracing the actions and seeing which action is completed and which not. Seeing the actions sequence will help us to establish and solve the problem we have. We count on your help for the solution of this problem. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QoS-aware cable/dsl routers?
Which cable/dsl routers on the market are QoS aware? I know about the linksys WRT54G with a hacked firmware and I have been looking at other routers' specs but no clear mention of the feature. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Docs
Kevin Walsh wrote:
Neil Cherry [EMAIL PROTECTED] wrote:
OK, this may seem to be an obvious question but where do I find
the reference docs? I'm getting this error message:
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know what the
parameters mean. I looking for a man page basically.
It has nothing to do with the 't' in your Dial().
The Dial() command docs can be found here:
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Ah, a key to the kingdom, thanks!
The predefined extension names list, including 't', can be found in
here:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
The 't' context is called when a timeout occurs. You could get rid
of the warning with the following:
exten = t,1,Hangup
That would simply hang up the line when a timeout is detected. You
could do anything you like in there, of course.
This page could be helpful too:
http://www.voip-info.org/wiki-Asterisk+cmd+ResponseTimeout
Thanks, it appears that I need to learn to use Wiki.
The WiKi is your friend.
So far it hasn't been very friendly. I tried to to find a document
I printed out (it printed poorly). When I entered the document's
title it fail to list that link. I actually found it via google
(weird). I guess I need to learn a new way to think for searching
for Asterisk info. I'll learn. :-)
My current set of problems are just configuration problems. I'm not
used to the commands, how they work and what they do. I accidently
figured out 't' after I got an error about no 'i' for invalid
extensions. Right now I'm wrestling with a SJPhone and the
Grandstream. Both have their own annoyances but I figure * will
be able to work around most of those.
BTW, let me say thanks. I don't want everyone to think I'm
just complaining. It's more frustration with the steep learning
curve.
--
Linux Home Automation Neil Cherry[EMAIL PROTECTED]
http://home.comcast.net/~ncherry/ (Text only)
http://linuxha.sourceforge.net/ (SourceForge)
http://hcs.sourceforge.net/ (HCS II)
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[Asterisk-Users] Cannot install module Bri-Stuff-0.0.2 zaphfc.ko does not exist.
Hello all, I complied the Bri-Stuff-0.0.2 zaphfc.c and it produced a zaphfc.o which is not compatible with the 2.6 kernel. I am using suse 9.1. I will try and understand the makefile better as it does mention the .ko file, in the mean time if anyone has any idea why the zaphfc.ko is not built then please let me know. Thanks. Ian Hailey.
[Asterisk-Users] Strange behavioir on a exten
Hi, I've got a strange behavior on one of my new extentions. This is what I got: extensions.conf with to #includes: extensions-manual.conf extensions-db.conf I have around 50 users and 100 contexts in these files. I reacently included one: (created from db) [marcela] include = Common include = VoiceMail include = marcela-extensions [marcela-extensions] ;; === Marcela Arana exten = 5913020,1,Dial(SIP/marcela_5913020,30,tr) I have also a DID for that exten: ;; === DID === exten = didNumber,1,Goto(marcela,3020,1) When I call the DID, I get an invalid extention message. -- Executing Goto(SIP/10.0.0.5-08420070, marcela|5913020|1) in new stack -- Goto (marcela,5913020,1) -- Sent into invalid extension '5913020' in context 'marcela' on SIP/10.0.0.5-08420070 -- Executing Playback(SIP/10.0.0.5-08420070, invalid) in new stack Any Ideas why? Thanks in advance -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] execute a context from cron
Michael George wrote: On Thu, Jul 01, 2004 at 03:58:25PM +0200, Manuel Wenger wrote: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job It looks like this is just a conditional Goto and it will not spontaneously start a flow in a context. What I need is something that will, at a given time, act just like we picked up an internal extension and dialed a sequence of numbers. Thanks! Then you definitely want to take a look a sample.call, use the cron job to create you own file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] simple AGI script
hello, does anybody have some agi script that can do following : when extension didn't pickup phone call, system send mail notify ( via sendmail ) to user mailbox with date, time and caller id ? ( something like missed call ) please can somebody help me with whis ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk -BEGIN GEEK CODE BLOCK- Version: 3.1 GS/E/B/PA/SS d+(++) s+:+ a C++$ ULS !P+++(---)$ L$ E++ W++ !N w(+++) !O() M++ V--() Y+$ PGP+ t- !5? X- !R !tv at b++() DI++ D+++ at G e+++ h(*) at r% --END GEEK CODE BLOCK-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS-aware cable/dsl routers?
I use the open source m0n0wall software running on Soekris Engineering net4501. Have also used Linksys BEFSR-81 with QoS, but the v1 hardware had a problem with random loss of wanlan connectivity. Reports are that v3,3 hardware does not. The firmware allows physical port based priority, ie hi/med/low priority to the jack leading to the * server. Or IP port based priority assignment, limited to 10 ports. Also had a Draytek Vigor 2900G for a short while. It has bandwidth limiting capability, and enforces VPN over wireless (very cool) but the firmware was new and pretty lame. Michael On Thu, 1 Jul 2004 10:03:03 -0500, spectro wrote: Which cable/dsl routers on the market are QoS aware? I know about the linksys WRT54G with a hacked firmware and I have been looking at other routers' specs but no clear mention of the feature. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 With us or against us isn't a policy worthy of a democratic superpower. -- Zbigniew Brzezinski, Former US National Security Advisor ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sccp to sip call signalling
Hi, How asterisk decides whether to do media relaying or not? For SIP I've found that canreinvite=yes allows me to use * only for signalling, RTP stream will flow between endpoints only. Are such things possible when calling from SCCP channel to SIP for example? SCCP to SCCP? Thanks in advance! -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
On Thu, 2004-07-01 at 11:00, Miroslav Nachev wrote: Hi, We have our own algorithm handling (dial plan) the calls and different events. When we receive an external call (from FXO), probably in consequence of our algorithm, some times the FXO port remains open and we could not establish the reason why the port is not closing. We were thinking a lot what might be the problem - for example we might forget to call the hang-up method somewhere in the script. Unfortunately we were not able to fix the problem. We came to the conclusion that the only way to establish where the mistake is, is to ask you for information about is there any log files, which could help us tracing the actions and seeing which action is completed and which not. Seeing the actions sequence will help us to establish and solve the problem we have. We count on your help for the solution of this problem. You speak of FXO, this makes me assume you are speaking of an analog POTS line. If so, then your next question is which side of the call did the actual hangup. If the non asterisk side did the hangup, does it provide disconnect supervision? If no disconnect supervision, can you get a tone pattern for busydetect or callprogress to detect those events. Maybe searching around for those few new terms I just used above will get you hooked up with previous threads to understand anything else you need. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't transfer with Zap and SPA-2000
It doesn't look like your using t or T in your Dial command. The Wiki on voip-info.org will explain those flags. On Wed, 2004-06-30 at 20:05 -0700, Seth Mattinen wrote: I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial(SIP/206-2c61, Zap/1/###) in new stack Currently on call: -- Called 1/### Press flash to place call on hold with SPA-2000: -- Hungup 'Zap/1-1' As soon as I press the flash button on my SPA-2000 connected phone, the zap channel hangs up and the call is disconnected. The same procedure works fine between SIP channels (flash to hold, or transfer). I'm hoping this is just a config problem and not a general defect, because it seems odd to me that I can't transfer calls when a zap channel is the other end of the call I want to transfer or place on hold. sip.conf: [206] type=friend username=206 secret=blah host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 zapata.conf: [channels] language=en signalling=fxs_ks usecallerid=no echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=2.5 txgain=7.0 busydetect=yes busycount=8 faxdetect=incoming context=inbound-analog1 channel = 1 context=inbound-analog2 channel = 2 -- Seth experientia docet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot install module Bri-Stuff-0.0.2 zaphfc.ko does not exist.
in zaphfc subdirectory: make linux26 and make sure you have a: ln -s /usr/src/linux-2.6.7 /usr/src/linux ln -s /usr/src/linux-2.6.7 /usr/src/linux-2.6 Like: 0 lrwxrwxrwx 1 root root 11 Jun 22 21:16 linux - linux-2.6.7 0 lrwxrwxrwx 1 root root 11 Jun 22 21:16 linux-2.6 - linux-2.6.7 4 drwxrwxr-x 19 500 500 4096 Jun 25 15:53 linux-2.6.7 asterisk wrote: Hello all, I complied the Bri-Stuff-0.0.2 zaphfc.c and it produced a zaphfc.o which is not compatible with the 2.6 kernel. I am using suse 9.1. I will try and understand the makefile better as it does mention the .ko file, in the mean time if anyone has any idea why the zaphfc.ko is not built then please let me know. Thanks. Ian Hailey. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] execute a context from cron
Hi! I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. Use DbPut() to store a flag, DbGet() to check the flag and then act upon it with GotoIf(). You can also use asterisk -rx to issue commands like database put to perform the DbPut() from a script, i.e. outside of extensions.conf. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
Obviously the less I spend the better. But if we have to, a few thousand more I guess. The problem I have is that this setup is more of a trial run. Once it works, I'm going to be cloning slightly smaller setups to 9 other cities. But they are pretty small, 1 or 2 lines and 2-4 phones in each location. I will only be using POTS lines in each location. The current setup works great besides the echo, and some of the information I've read point to the Telco being the issue. If thats the case, I should in theory be able to get them to fix the problem. (though I could be dreaming) On Wed, 2004-06-30 at 22:42 -0500, Daniel Jimenez wrote: Mike Benoit wrote: So whats the next step? How much money are you willing to put in the project? Are you talking POTS lines or a PRI? If this is a serious project and you'd really like to clear it up I'd look at a Cisco device (maybe one of the newer rackmount 1700s) with FXO ports or a Serial interface for PRI. You can use h.323 or SIP to communicate with the device. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration failed for SIP
Yes. Remove the default IP. Kurt __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS-aware cable/dsl routers?
Using the sveasoft firmware on a WRT54GS (newer version of the G, about $20 more) the QOS feature does work. It's not quite as robust as a corporate-grade router, but it does work well for me, ensuring that I set aside 100k of bandwidth any time my VoIP box is running. On Thu, 1 Jul 2004 10:03:03 -0500, spectro [EMAIL PROTECTED] wrote: Which cable/dsl routers on the market are QoS aware? I know about the linksys WRT54G with a hacked firmware and I have been looking at other routers' specs but no clear mention of the feature. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
I've got the same problem NEAR end echo (We hear the echo on OUR side, person on the PSTN never hears it..) We're tyring to get our PRI carrier to run us through a echo can, or re-write it through a switch they have which has built in echo cans... Ugg.. Thanks, Billy - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 10:56 AM Subject: Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next? On Thursday 01 July 2004 08:39, Robinson Tim-W10277 wrote: All voip providers will use digital 4-wire interconnect to Asterisk or similar, so echo problems are much reduced, as there are only 'echo points' at the far end and your handset. And on my PRI that is specifically where my echo is coming from... the far end. VOIP calls through nufone have no echo MOST PSTN calls through the PRI have no echo SOME PSTN calls (usually to local numbers NOT terminated at my local CO) have significant echo... I too have been unsuccessful in getting this zapped. My connection: Norstart MICS -- Adit600 --- T100P -- IAX2 -- TE405P -- Bell Canada PRI *1 = Xeon/2.4 with HT with T100P *2 = Xeon/2.4 with HT with TE405P *2 also does the NuFone IAX2 connection (it is always in the loop, as *1 is on a private network) Strange stuff, I am going to look at T1 echo cancellation hardware if I cant' get this solved. Tried: - echotraining=800 on *1 and *2 - echocancel=32,64,128 on both Eventually the MICS will have a digital connection to *1 instead of going through the Adit600 but we haven't got there yet :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't transfer with Zap and SPA-2000
No t or T needed it works fine if you use ulaw. I do it every single day. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Benoit Sent: Thursday, July 01, 2004 10:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can't transfer with Zap and SPA-2000 It doesn't look like your using t or T in your Dial command. The Wiki on voip-info.org will explain those flags. On Wed, 2004-06-30 at 20:05 -0700, Seth Mattinen wrote: I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial(SIP/206-2c61, Zap/1/###) in new stack Currently on call: -- Called 1/### Press flash to place call on hold with SPA-2000: -- Hungup 'Zap/1-1' As soon as I press the flash button on my SPA-2000 connected phone, the zap channel hangs up and the call is disconnected. The same procedure works fine between SIP channels (flash to hold, or transfer). I'm hoping this is just a config problem and not a general defect, because it seems odd to me that I can't transfer calls when a zap channel is the other end of the call I want to transfer or place on hold. sip.conf: [206] type=friend username=206 secret=blah host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 zapata.conf: [channels] language=en signalling=fxs_ks usecallerid=no echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=2.5 txgain=7.0 busydetect=yes busycount=8 faxdetect=incoming context=inbound-analog1 channel = 1 context=inbound-analog2 channel = 2 -- Seth experientia docet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound: Record Overrun
Hi, When I dial into asterisk I set it up in extensions.conf so it will play some messages, but when I dial in asterisk picks up but I hear no sound. There is moments of silence where the audio should be playing but I get nothing. I checked /var/log/messages to see what was wrong and I got the following error: Jun 29 20:46:33 eclipse kernel: Sound: Recording overrun Does this mean the computer that asterisk is running on gets simply too bogged down and can't process sound while asterisk is running (because it can play audio when asterisk isn't running) or is it something else? It is a pentium 133 running with Redhat 8 and I have a soundblaster sound card by the way. Thanks for any help I get. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] execute a context from cron
Why don't you include a context on a schedule? I have a support queue included only at certain times, such as monday through friday, 6a to 6p. All other times I include a context that sends that extension straight to voicemail. Check out http://www.voip-info.org/wiki-Asterisk+tips+openhours On Jul 1, 2004, at 6:46 AM, Michael George wrote: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. I figured that one way to do it would be to have cron run asterisk -rxsome command if there were some command that would tell asterisk to go to a specific context,extension,priority, but I cannot find that command. Does such a command exist? Or is there a better way to do this that I have overlooked? Thank you! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Docs
On 08:10 AM 7/1/2004, Neil Cherry wrote: The WiKi is your friend. So far it hasn't been very friendly. I tried to to find a document I printed out (it printed poorly). When I entered the document's title it fail to list that link. I actually found it via google (weird). I guess I need to learn a new way to think for searching for Asterisk info. I'll learn. :-) The Wiki's search system seems to leave a lot to be desired. I don't often use the search system in the wiki, I tend to either 'google' it, or if I know what I'm looking for, I generally navigate manually to the page in the wiki. However, this assumes you've spent a good bit of time working with the wiki at www.voip-info.org and know what you are looking for and how it's going to be titled. A few days ago I was looking for some info from the asterisk-addon cdr_odbc and I entered 'Asterisk cdr mysql. Well the search engine found the page I was looking for, and it was titled Astersk cdr mysql, and yet, even though it was a perfect match, it was around #5 in the results. *shrug* references to pages of the voip-info.org wiki from google is your friend. There are also quite a few friends you can buy on asterisk-biz list, if you are so inclined. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
On Thursday 01 July 2004 12:03, Billy Huddleston wrote: I've got the same problem NEAR end echo (We hear the echo on OUR side, person on the PSTN never hears it..) That's what I have -- I called it far-end echo because I hear the echo, so it's coming from the far-end. We're tyring to get our PRI carrier to run us through a echo can, or re-write it through a switch they have which has built in echo cans... Bell claims they have no idea what a T1 echo canceller is. hahaha. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
Hello Steven, The caller (originator, PSTN side) is closed the line, but the asterisk side can't understand that the caller is Hangup the line. Our PSTN is based on Siemens and Ericsson. I found some materials (documentation of Siemens PBX) where the process of negotiation is described (tones in Hz, times, etc.) but I don't know how to enter this data in Asterisk files. There is not description for this information. Also, I am looking for Caller ID detection. If you can help me will be very good. I try UK settings, but this is not working in Bulgaria. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Thursday, July 1, 2004, 6:28:52 PM, you wrote: SC On Thu, 2004-07-01 at 11:00, Miroslav Nachev wrote: Hi, We have our own algorithm handling (dial plan) the calls and different events. When we receive an external call (from FXO), probably in consequence of our algorithm, some times the FXO port remains open and we could not establish the reason why the port is not closing. We were thinking a lot what might be the problem - for example we might forget to call the hang-up method somewhere in the script. Unfortunately we were not able to fix the problem. We came to the conclusion that the only way to establish where the mistake is, is to ask you for information about is there any log files, which could help us tracing the actions and seeing which action is completed and which not. Seeing the actions sequence will help us to establish and solve the problem we have. We count on your help for the solution of this problem. SC You speak of FXO, this makes me assume you are speaking of an analog SC POTS line. SC If so, then your next question is which side of the call did the actual SC hangup. If the non asterisk side did the hangup, does it provide SC disconnect supervision? If no disconnect supervision, can you get a tone SC pattern for busydetect or callprogress to detect those events. SC Maybe searching around for those few new terms I just used above will SC get you hooked up with previous threads to understand anything else you SC need. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zultys 4x4 or 4x5 ip phones?
Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Meetings are indispensable when you don't want to do anything - John Kenneth Galbraith ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip to Sip
I appologize if this was already answered somwhere onhttp://www.voip-info.org/wiki-Asterisk, I'm sure it probably is.And if you wish to just point me to a link that would be appreciated. I am very new to asterisk and unix all around, so these questions maysound rather ignorant. First being, how do I setup asterisk to point to another asterisk server and make all the lines which should be PSTN or POTS go directly to another existing asterisk server by using accounts? For instance, if I was using another asterisk service with my voip phone to connect to it how could I make my local server use that account as a line? Also, Is there any really good documentation on configuring asterisk, besides the asterisk handbook. Maybe something a little more indepth and something explaining all the commands available on the console? Thanks a lot -chad
Re: [Asterisk-Users] execute a context from cron
I simply use ; Timing list for includes is ; ; time range|days of week|days of month|months ; include = day|09:30-17:45|mon-fri|*|* include = eve|17:45-23:00|mon-fri|*|* include = eve|00:00-23:59|sat-sun|*|* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zultys 4x4 or 4x5 ip phones?
Michael Graves wrote: Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael I have used the 4x4, it worked very well with asterisk. We are going to be buying a few for the office when we get * implemented. We found a local company that sells them and they loaned us one for 3 weeks to test. Kyle www.quadrasoftware.com Asterisk Call management applications. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] execute a context from cron
On Thu, Jul 01, 2004 at 09:13:58AM -0700, Chad Scott wrote: Why don't you include a context on a schedule? I have a support queue included only at certain times, such as monday through friday, 6a to 6p. All other times I include a context that sends that extension straight to voicemail. Check out http://www.voip-info.org/wiki-Asterisk+tips+openhours Yes, openhours was easy. What I need is to *initiate* a call at a certain time. I want to have asterisk pick up the line and enter the key sequence to have the POTS do the forwarding, not asterisk. And then when we open up again I want it to call the POTS and turn off the call forwarding. I think the sample.call will do what I want, though. I'll look into that. Thanks! I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. I figured that one way to do it would be to have cron run asterisk -rxsome command if there were some command that would tell asterisk to go to a specific context,extension,priority, but I cannot find that command. Does such a command exist? Or is there a better way to do this that I have overlooked? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DISA and AGI: authenticate by caller ID?
I'm having trouble getting an AGI exec command to spawn app_disa. The
script executes properly, but does not spawn DISA. The CLI gives no helpful
clues. Am I doing the exec incorrectly?
I want to have a way to authenticate callers to the extension by Caller
ID... if their caller ID is in my database and set to active, they can call
out. [like a calling card but auth'd by CID instead of PIN].
Here is my dialplan:
1234, 1, agi(ldusers.agi)
1234, 2, Hangup
Here is my code:
#!/usr/bin/perl
#
use Asterisk::AGI;
use DBI;
$db = dbname;
$host = hostname;
$port = 3306;
$userid = dbuser;
$password = dpasswd;
$connectionInfo = DBI:mysql:database=$db;$host:$port;
$dbh = DBI-connect($connectionInfo,$userid,$password);
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
if (my $callerid = $input{'callerid'}) {
$AGI-say_digits($callerid);
$query = SELECT active FROM cids WHERE cid=$callerid;#
active should be 1 if the caller ID is found and set active
$sth = $dbh-prepare($query);
$sth-execute();
$sth-bind_columns(undef, \$active);
$sth-fetch();
if($active)
$AGI-exec('DISA','no-password|disa');
}
$AGI-hangup();
exit;
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RE: [Asterisk-Users] Zultys 4x4 or 4x5 ip phones?
[EMAIL PROTECTED] wrote: Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael Im looking at purchasing some of these phones as well...But with how poor their sales team have been so far, im beginning to wonder. I emailed them on Monday for info along with a reseller in my region, and have called their sales team numerous times only to be told that 1 person handles my region and for some reason he never answers the phone or returns calls. Its been 4 days that ive been trying to buy some of their phones, with no luck.Id purchase them online, but not at their list price when their resellers give 10% discounts. --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zultys 4x4 or 4x5 ip phones?
[EMAIL PROTECTED] wrote: Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael Oh and i see your in the same region as myself in houstongood luck =) let me know if you have success with them, for im still interested in their quality of phones. --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Docs
Chris A. Icide wrote: On 08:10 AM 7/1/2004, Neil Cherry wrote: The WiKi is your friend. So far it hasn't been very friendly. I tried to to find a document I printed out (it printed poorly). When I entered the document's title it fail to list that link. I actually found it via google (weird). I guess I need to learn a new way to think for searching for Asterisk info. I'll learn. :-) The Wiki's search system seems to leave a lot to be desired. I don't often use the search system in the wiki, I tend to either 'google' it, or if I know what I'm looking for, A few days ago I was looking for some info from the asterisk-addon cdr_odbc and I entered 'Asterisk cdr mysql. Well the search engine found the page I was looking for, and it was titled Astersk cdr mysql, and yet, even though it was a perfect match, it was around #5 in the results. *shrug* references to pages of the voip-info.org wiki from google is your friend. I'm very good with google, I have it setup so I can type the search list into the url bar (or g and the list for things that look like a URL). There are also quite a few friends you can buy on asterisk-biz list, if you are so inclined. Nah, I'm actually trying to learn VoIP (yes the entire thing) and paying someone to do it won't help me learn. It's got to be learned by doing and search if you really want to know it. When I ask questions here I prefer pointers so I can learn to 'fish' so I can 'feed' myself. :-) -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
reseaux: update to the latest CVS version (from /usr/src, do cvs update asterisk), and give me an email address I can send a patched indications.c to. Then put the file I send you in /usr/src/asterisk (overwrite the existing indications.c), and do make; make install as root (without asterisk running). That should stop the bug from killing your calls, but after talking to bkw_, I think the bugfix may just cover up a larger bug. By the way, would you happen to be using BroadVoice or another SIP provider for Asterisk? Holger: I'm going to file a bug report today. I asked a few questions about it yesterday while talking to bkw_; just didn't have time to do the report yesterday. At 04:17 AM 7/1/2004, you wrote: Dear Ted i have notice the same problem had you reported from monday, i have try to update to today CVS HEAD but nothing still buggy so i roolback to Stable V1. Where i can find the pacth? Thanks in advance Dimitri On Wednesday 30 June 2004 08:39 pm, programmer_ted wrote: Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted [EMAIL PROTECTED] P.S. Read to the very end. The original bugfix has an annoying side effect. Hi, My friend and I were getting a warning when calling his Sipura from a PSTN line (connecting to Asterisk through BroadVoice), that said: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) and was followed by a hangup (type 64 is 16-bit Signed Linear PCM, type 4 is G711u). I found that many people have had similar issues, but these were never resolved. So, I figured that because Asterisk is open-source, I'd dive into the code and try to fix the bug. After a couple of hours of familiarizing myself with the Asterisk code and tracing the problem, I found that for some reason the tone generator, which uses 16-bit Signed Linear PCM, was still being allocated and playtones_generator (indications.c) was still getting called, regardless that the Sipura doesn't take SLINEAR data (in my case, it accepts G711u). So, I ended up adding an if conditional to the beginning of the playtones_alloc function (indications.c) to check if SLINEAR was supported by the channel, and if not, return 0 (which, when received by the ast_activate_generator function (channel.c), causes the channel generatordata to remain empty, effectively stopping the SLINEAR data from being sent in the most nonintrusive way possible). NOTICE: this bugfix will work for similar issues involving format 64 (16-bit Signed Linear PCM) being sent even if channel capabilities don't allow it, if the generator is involved - it's not limited to my situation (dialing the Sipura from Asterisk). This patch should be applied to indications.c under the main asterisk source directory (usually /usr/src/asterisk): 68a69 if (!(chan-nativeformats AST_FORMAT_SLINEAR)) return 0; Oh, and finally, here's a shameless plug to a good friend's website (which includes a VOIP forum!): http://outcast.ws Comments? Questions? :) Just a quick update. I was looking things over again and it appears this fix also disables the generator when I'm calling in on PSTN over BroadVoice (when dialing the Sipura), not just disabling it for the Sipura. This basically disables the dialing sound while waiting for the Sipura to pick up. I have an idea that I should have used chan-capabilities rather than chan-nativeformats, but it's too late to check at the moment. I'll try it out first thing tomorrow and update you guys, but for now, that's one drawback of using this fix. I thought it over a little bit more and the optimum solution would be to just translate the SLINEAR data to a format that is recognized by whoever is receiving the data, thus eliminating all drawbacks. I'm going to try using capabilities rather than nativeformats as a quick workaround (after debugging to see if it'll work), and then work on adding the translating code to sip_write. Actually, thinking about it again, it'd probably be best to just translate at the playtones_generator function. I'll keep you guys updated. ...snipped non-relevant signature info etc... Learning as I go. It appears I don't have access to the capabilities value from the ast_channel structure. I'm just gonna go ahead and have the SLINEAR data translate to the channel's writeformat. Ok, as I thought, PSTN over BroadVoice does not understand SLINEAR natively, which is why the dialing sound was also disabled when I dialed the Sipura. I added some code to playtones_alloc (indications.c) so that the write format is only set to SLINEAR if it's supported, and added some code to playtones_generator to translate from
Re: Re[2]: [Asterisk-Users] How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
On Thu, 2004-07-01 at 11:34, Miroslav Nachev wrote: Hello Steven, The caller (originator, PSTN side) is closed the line, but the asterisk side can't understand that the caller is Hangup the line. Our PSTN is based on Siemens and Ericsson. I found some materials (documentation of Siemens PBX) where the process of negotiation is described (tones in Hz, times, etc.) but I don't know how to enter this data in Asterisk files. There is not description for this information. Go look at busydetect and callprogress, There is documentation on the process. I don't need it and don't want to learn it so I can spoon feed others. Also, I am looking for Caller ID detection. If you can help me will be very good. I try UK settings, but this is not working in Bulgaria. Do you know what type of caller id your system is sending? That would be a good first step. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel wont compile errors on zttest
Hi all, I'm unable to complete the installation of zaptel. I did a checkout today from cvs. When I do make clean; make install. It gives me an error at this portion of the installation: /include/linux/modversions.h -DSTANDALONE_ZAPATA -c wct4xxp.ccc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o ztcfg.o ztcfg.ccc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -o zonedata.lo zonedata.ccc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -o tonezone.lo tonezone.car rcs libtonezone.a zonedata.lo tonezone.locc -o ztcfg ztcfg.o -lm -L. libtonezone.acc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o torisatool.o torisatool.ccc -o torisatool torisatool.occ -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o ztmonitor.o ztmonitor.ccc -o ztmonitor ztmonitor.occ -c ztspeed.ccc -o ztspeed ztspeed.occ -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o zttool.o zttool.ccc -o zttool zttool.o -lnewtcc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA zttest.c -o zttestzttest.c: In function `main':zttest.c:65: parse error before ')' tokenmake: *** [zttest] Error 1 I am running Red Hat Linux, Kernel is: linux-2.4.20-8 Any ideas? TIA Jon
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
On Thu, 1 Jul 2004, Mike Benoit wrote: Obviously the less I spend the better. But if we have to, a few thousand more I guess. The problem I have is that this setup is more of a trial run. Once it works, I'm going to be cloning slightly smaller setups to 9 other cities. But they are pretty small, 1 or 2 lines and 2-4 phones in each location. I totally understand this. My users complain frequently about echo, and I am still unable to determine why sometimes it works great, other's it does not. The CPU and Memory are powerful enough to handle it, and we rarely ever see any load on the box. I too feel this is the major caveat to Asterisk right now. I am curious how anyone is achieving a near echo free system. We are shooting for 1 out of every 300 calls to have echo, which I think can be a realistic goal. Given the nature of open source, and the mix-and-match of components that come up, I can see where Digium is in a hard place to nail down the cause of every occurance. I will only be using POTS lines in each location. The current setup works great besides the echo, and some of the information I've read point to the Telco being the issue. If thats the case, I should in theory be able to get them to fix the problem. (though I could be dreaming) I think ultimately, if a Mediatrix box, or Cisco box can accomplish echo cancellation, Asterisk should be able to do it with as much success. Being that I am not an experienced Programmer, I try not to complain to loudly. With my level of involvement, I typically make the business case to customers and spec out ROI, etc. I do have a technical background, and am getting better at trouble shooting Asterisk and working on the source code. In fact, subscribing to the CVS list has taken me leap years ahead of understanding the changes and why they are being committed. I don't know how much more putting a DSP to handle echo can on the cards would cost, but if it were 400 - 500 more I would certainly pay it without a second thought, provided it worked. Echo, I think, is the largest draw back to VoIP, and will be the limit to entry into many businesses. I know my client, if they were to do it all over again, would choose a regular TDM (nortel, avaya) solution over the echo they are experiencing. I think asterisk is definitly headed in the right direction though, and nothing good comes over night. So everyone who has worked on it deserves to be commended. Without their insight and dedication, we wouldn't even be talking about this, or have alternatives to turn to. Regards, - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
Ya I tried to duplicate this problem but couldn't... So It think it's a problem elsewhere but we shall see once mark looks at it. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of programmer_ted Sent: Thursday, July 01, 2004 12:19 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45 reseaux: update to the latest CVS version (from /usr/src, do cvs update asterisk), and give me an email address I can send a patched indications.c to. Then put the file I send you in /usr/src/asterisk (overwrite the existing indications.c), and do make; make install as root (without asterisk running). That should stop the bug from killing your calls, but after talking to bkw_, I think the bugfix may just cover up a larger bug. By the way, would you happen to be using BroadVoice or another SIP provider for Asterisk? Holger: I'm going to file a bug report today. I asked a few questions about it yesterday while talking to bkw_; just didn't have time to do the report yesterday. At 04:17 AM 7/1/2004, you wrote: Dear Ted i have notice the same problem had you reported from monday, i have try to update to today CVS HEAD but nothing still buggy so i roolback to Stable V1. Where i can find the pacth? Thanks in advance Dimitri On Wednesday 30 June 2004 08:39 pm, programmer_ted wrote: Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted [EMAIL PROTECTED] P.S. Read to the very end. The original bugfix has an annoying side effect. Hi, My friend and I were getting a warning when calling his Sipura from a PSTN line (connecting to Asterisk through BroadVoice), that said: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) and was followed by a hangup (type 64 is 16-bit Signed Linear PCM, type 4 is G711u). I found that many people have had similar issues, but these were never resolved. So, I figured that because Asterisk is open-source, I'd dive into the code and try to fix the bug. After a couple of hours of familiarizing myself with the Asterisk code and tracing the problem, I found that for some reason the tone generator, which uses 16-bit Signed Linear PCM, was still being allocated and playtones_generator (indications.c) was still getting called, regardless that the Sipura doesn't take SLINEAR data (in my case, it accepts G711u). So, I ended up adding an if conditional to the beginning of the playtones_alloc function (indications.c) to check if SLINEAR was supported by the channel, and if not, return 0 (which, when received by the ast_activate_generator function (channel.c), causes the channel generatordata to remain empty, effectively stopping the SLINEAR data from being sent in the most nonintrusive way possible). NOTICE: this bugfix will work for similar issues involving format 64 (16-bit Signed Linear PCM) being sent even if channel capabilities don't allow it, if the generator is involved - it's not limited to my situation (dialing the Sipura from Asterisk). This patch should be applied to indications.c under the main asterisk source directory (usually /usr/src/asterisk): 68a69 if (!(chan-nativeformats AST_FORMAT_SLINEAR)) return 0; Oh, and finally, here's a shameless plug to a good friend's website (which includes a VOIP forum!): http://outcast.ws Comments? Questions? :) Just a quick update. I was looking things over again and it appears this fix also disables the generator when I'm calling in on PSTN over BroadVoice (when dialing the Sipura), not just disabling it for the Sipura. This basically disables the dialing sound while waiting for the Sipura to pick up. I have an idea that I should have used chan-capabilities rather than chan-nativeformats, but it's too late to check at the moment. I'll try it out first thing tomorrow and update you guys, but for now, that's one drawback of using this fix. I thought it over a little bit more and the optimum solution would be to just translate the SLINEAR data to a format that is recognized by whoever is receiving the data, thus eliminating all drawbacks. I'm going to try using capabilities rather than nativeformats as a quick workaround (after debugging to see if it'll work), and then work on adding the translating code to sip_write. Actually, thinking about it again, it'd probably be best to just translate at the playtones_generator function. I'll keep you guys updated.
RE: [Asterisk-Users] cisco phone and parked calls
http://65.38.28.146/app_valetparking.c
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joe Antkowiak
Sent: Thursday, July 01, 2004 12:42 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cisco phone and parked calls
hmmm... Where can I get this?
On Thu, 1 Jul 2004 00:29:29 -0500, Brian K. West [EMAIL PROTECTED] wrote:
Two words...
Valet Parking...
bkw
- Original Message -
From: Joe Antkowiak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 01, 2004 12:25 AM
Subject: Re: [Asterisk-Users] cisco phone and parked calls
Does anyone have any input on this? I tried using what Craig said
above, but it didn't work...
On Wed, 30 Jun 2004 13:02:46 -0400, Joe Antkowiak [EMAIL PROTECTED]
wrote:
So, in order to use the parking extension configured in
parking.conf,
I have to configure that extension under a [parkedcalls] context in
my
extensions.conf? I thought the call parking app was supposed to
take
care of that for me?
On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
[EMAIL PROTECTED] wrote:
In my sip extensions context I have
include = parkedcalls
In extensions.conf I have
[parkedcalls]
Exten = 2000,1,Answer
In parking.conf I have the same.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Antkowiak
Sent: 29 June 2004 22:56
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco phone and parked calls
sent this before, but it bounced back and didn't show up on the
list.
If it did get sent, I apologize.
-- Forwarded message --
From: Joe Antkowiak [EMAIL PROTECTED]
Date: Tue, 29 Jun 2004 14:55:25 -0400
Subject: cisco phone and parked calls
To: [EMAIL PROTECTED]
So, I can't figure out how to get the parkandannounce application
to
work the way I want it to... I have cisco 7960 IP phones using
SIP,
and this is what I have in my extensions.conf:
exten =
700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
XTEN:1},1)
exten = 700,2,Hangup
and in my parking.conf:
[general]
parkext = 700 ; What ext. to dial to
park
parkpos = 701-720 ; What extensions to park
calls
on
context = parkedcalls ; Which context parked
calls
are
in
parkingtime = 180
In order for the person parking the call to hear what parked
extension
the call is on, they have to do the transfer by pressing # and
dialing
700. When the user uses the transfer function on the cisco phone,
it
still correctly parks the call, but never tells the person what
extension its parked on.
Also, for some reason, I had to create that 700 extension, it
always
complains that it can't find 700 when I don't do that, even though
parkedcalls is included in the internal context...
Any suggestions?
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--
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antkojm1 (at) gmail.com
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Re: [Asterisk-Users] cisco phone and parked calls
h... Is there any way to make it say the parking lot space a call
is being parked into, on the channel that is calling into the
extension that is running the ValetParkCall app? My customer wants to
know what space it is without having to listen to all the parked
calls, and uses attended transfer...
On Thu, 1 Jul 2004 12:40:35 -0500, brian [EMAIL PROTECTED] wrote:
http://65.38.28.146/app_valetparking.c
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joe Antkowiak
Sent: Thursday, July 01, 2004 12:42 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cisco phone and parked calls
hmmm... Where can I get this?
On Thu, 1 Jul 2004 00:29:29 -0500, Brian K. West [EMAIL PROTECTED] wrote:
Two words...
Valet Parking...
bkw
- Original Message -
From: Joe Antkowiak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 01, 2004 12:25 AM
Subject: Re: [Asterisk-Users] cisco phone and parked calls
Does anyone have any input on this? I tried using what Craig said
above, but it didn't work...
On Wed, 30 Jun 2004 13:02:46 -0400, Joe Antkowiak [EMAIL PROTECTED]
wrote:
So, in order to use the parking extension configured in
parking.conf,
I have to configure that extension under a [parkedcalls] context in
my
extensions.conf? I thought the call parking app was supposed to
take
care of that for me?
On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
[EMAIL PROTECTED] wrote:
In my sip extensions context I have
include = parkedcalls
In extensions.conf I have
[parkedcalls]
Exten = 2000,1,Answer
In parking.conf I have the same.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Antkowiak
Sent: 29 June 2004 22:56
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco phone and parked calls
sent this before, but it bounced back and didn't show up on the
list.
If it did get sent, I apologize.
-- Forwarded message --
From: Joe Antkowiak [EMAIL PROTECTED]
Date: Tue, 29 Jun 2004 14:55:25 -0400
Subject: cisco phone and parked calls
To: [EMAIL PROTECTED]
So, I can't figure out how to get the parkandannounce application
to
work the way I want it to... I have cisco 7960 IP phones using
SIP,
and this is what I have in my extensions.conf:
exten =
700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
XTEN:1},1)
exten = 700,2,Hangup
and in my parking.conf:
[general]
parkext = 700 ; What ext. to dial to
park
parkpos = 701-720 ; What extensions to park
calls
on
context = parkedcalls ; Which context parked
calls
are
in
parkingtime = 180
In order for the person parking the call to hear what parked
extension
the call is on, they have to do the transfer by pressing # and
dialing
700. When the user uses the transfer function on the cisco phone,
it
still correctly parks the call, but never tells the person what
extension its parked on.
Also, for some reason, I had to create that 700 extension, it
always
complains that it can't find 700 when I don't do that, even though
parkedcalls is included in the internal context...
Any suggestions?
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Joe Antkowiak
antkojm1 (at) gmail.com
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[Asterisk-Users] Pager Notification
Hi; Before I tell a customer that this would require custom development I figured I would ask here. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Best Wishes. Chris Travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard
RE: [Asterisk-Users] execute a context from cron
Michael George [EMAIL PROTECTED] wrote: What I need is to *initiate* a call at a certain time. I want to have asterisk pick up the line and enter the key sequence to have the POTS do the forwarding, not asterisk. And then when we open up again I want it to call the POTS and turn off the call forwarding. I think the sample.call will do what I want, though. I'll look into that. This will help: http://www.voip-info.org/wiki-Asterisk+auto-dial+out You can schedule into the future by setting the call file's modification time accordingly. If it's to be a recurring job then cron would be good, as you said. Perhaps you should consider a Sipura SPA-2000 or similar at your forward-to location and let Asterisk handle the forwarding. I just put my phone on DnD and let the answering machine take messages. I don't want people to be able to reach me any time of the day or night. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DISA and AGI: authenticate by caller ID?
Hi Matthew,
Look at the bootom for my recommendation (take note, I did not test it):
On Thu, 2004-07-01 at 14:08, Matthew Simpson wrote:
I want to have a way to authenticate callers to the extension by Caller
ID... if their caller ID is in my database and set to active, they can call
out. [like a calling card but auth'd by CID instead of PIN].
Here is my dialplan:
1234, 1, agi(ldusers.agi)
1234, 2, Hangup
Here is my code:
#!/usr/bin/perl
#
use Asterisk::AGI;
use DBI;
$db = dbname;
$host = hostname;
$port = 3306;
$userid = dbuser;
$password = dpasswd;
$connectionInfo = DBI:mysql:database=$db;$host:$port;
$dbh = DBI-connect($connectionInfo,$userid,$password);
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
if (my $callerid = $input{'callerid'}) {
$AGI-say_digits($callerid);
$query = SELECT active FROM cids WHERE cid=$callerid;#
active should be 1 if the caller ID is found and set active
$sth = $dbh-prepare($query);
$sth-execute();
$sth-bind_columns(undef, \$active);
$sth-fetch();
if($active)
$AGI-exec('DISA','no-password|disa');
^
Instead of executing the application, try creating a new context in your
dialplan that executes DISA. You can send the call to that context like
this:
$AGI-set_context(disa);
$AGI-set_extension(s);
$AGI-set_priority(1);
}
$AGI-hangup();
exit;
In extension.conf add the disa context like this:
[disa]
exten = s,1,disa,no-password|disa
This way, if an error happens with DISA, it will be displayed at the
asterisk console (it will not be hidden inside AGI).
Good luck,
--
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.
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Re: [Asterisk-Users] voicemail notification?
Hi Rich, On Thu, 2004-07-01 at 11:36, Rich Adamson wrote: Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a second email address (eg, pager), it works but the first address does not, like: 3002 = 3002,Rich,[EMAIL PROTECTED],[EMAIL PROTECTED] Played with the context to ensure that wasn't an issue. Faintly remember seeing something modified via cvs list, but can't seem to find anything addressing this one. Google doesn't provide any hints. Thoughts? Another bug was introduced in function notify_new_message: the event sent to manager does not include the voicemail context, so the manager notifications allways return 0 messages. I will submit a bug/patch to the bugtracker for this (as it affects the MWI in my flash operator panel), and I will try to look also at your problem. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pager Notification
Chris Travers Wrote Before I tell a customer that this would require custom development I figured I would ask here. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? For our notifications, we just send e-mails as text messages to their cell phones. Most of our users have cell phones, and with the invent of SMS and text messaging, pagers are no longer needed. For verizon, we shoot emails to [EMAIL PROTECTED] and for Att customers, we shoot e-mails to [EMAIL PROTECTED] Both work great. Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Not sure here, but I am sure there is, with some AGI scripting or using the sample.call file. You can then run Background to await DTMF input. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pager Notification
On Thu, 2004-07-01 at 13:03, Chris Travers wrote: Hi; Before I tell a customer that this would require custom development I figured I would ask here. These questions point to you not being qualified to have customers yet. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? AGI isn't the route for this. Most pagers support an email gateway, just use it. Maybe you need to trigger it with a procmail rule. Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? sample.call You may wish to call a real consultant to bail you out. Don't bother calling me, I'm not a consultant. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zultys 4x4 or 4x5 ip phones?
On Thu, 2004-07-01 at 10:51, Michael Graves wrote: Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? We've been using the 4x4's extensively for a while now. Out of all of the phones we've tried they've sounded the best and had the best overall feature set. We've been selling them to all of our customers for all of their high-volume users. We should be getting some 4x5's in shortly for testing. From what I've seen so far they seem pretty much identical to the 4x4's except for the extra NAT/VPN/Firewall features and the Bluetooth stuff. From what I know of them so far I don't BELIEVE that they will forward PSTN calls in any way. The line terminates on that handset and that's the end of it. Don't quote me on that though. :) -- Alex Malinovich Golden Technologies, Inc. (219) 462-7200 x 216 http://www.golden-tech.com signature.asc Description: This is a digitally signed message part
