Re: [asterisk-users] BLF with SPA941?
- Original Message - From: Ex Vito To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 3:59 PM Subject: Re: [asterisk-users] BLF with SPA941? Although I've never tested such feature on those devices, I know that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?). Are you running it ? Appearently, the latest firmware for SPA9x1 is 5.1.8, the SPA9x1 is not receiving the 5.2. and 6.1 firmware. There is a somewhat heated discussion here, which I unfortunately didn't read before ordering. https://www.myciscocommunity.com/thread/1541 SPA9X1 are our entry-level business IP Phones. Their feature set will not evolve (no new features are expected on this series) from what's existing today (5.1.8). For LDAP and other features we recommend the SPA9X2 product family, which is our mainstream SIP Small Business IP phones family. I am still thinking this i a political decision, and the SPA9x1 would be able to support the BLF etc features. Reminds me of a rumour that an upgrade to some photocopiers to offer more features, costing several k$, were simply a technician cutting a jumper; the hardware were already ready. But I'm not ready to try forcing a SPA9x2-software into a SPA941. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
hi all, i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE GUYS UR HELP N SUPPORT WILL BE HIGHLY APPRECIATED. THX___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
I think just renaming the [default] to [public] or [unautorized], and a comment saying Don't put outgoing calls in this context, as unauthorized users, even from outside, are routed here by default. would be enough. I'm not sure if local phones should automatically be routed to a [local] context. I think the [public] should be available for guest users, and be published, or at least be in the enum database. Why should my call (and my money) go from my desk via my ip-pabc to my voisp possibly through pstn (through echelon) to your voisp to your ip-pabc to your desk, when it could go from my ip-pabc to your ip-pabc directly. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
- Original Message - From: aster...@opensourcesolution.in To: asterisk-users@lists.digium.com Sent: Friday, November 13, 2009 9:47 AM Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE hi all, i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE guys ur help n support will be highly appreciated. There are many fine explanations on the net. Read and try, if you then have problems with the details, come back. Or you can pay a consultant to do your work Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call Ring
Is there any documentation on the CallWaitingRing? Thanks Dan -Original Message- From: Danny Nicholas da...@debsinc.com Sent: 12 November 2009 14:21 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming Call Ring Depending on your phone, you can use CallWaitingRing to ring the phone anyway. I do this with Polycom 501s. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, November 12, 2009 6:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming Call Ring Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is still ringing, that extension doesnt ring. This is because when the Dial command was executed, that extension was busy. Is there any way to make that extension ring as soon as its available if there is still an incoming call? Thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
On 13 Nov 2009, at 08:47, aster...@opensourcesolution.in aster...@opensourcesolution.in wrote: I had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE guys ur help n support will be highly appreciated. thx What is it with you! ARGH. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] little boy on asterisk and Debian
Hi Everybody, Sorry for my bad english I'm french! I need your help! I installed asterisk1.6.1 with update 1.6.9, Dahdi for musiconhold and chatroom. I created sip users, and a IVR. I can use my webradio to put in musiconold. That's good but the music is cutted. I would like to know how can I configure chan_dahdi.conf? Must I create Channels? Can you help me please? Thank you very much. _ AMICALEMENT... Manu - Sites Web : http://www.manu-dpk.net WebRadio : http://www.manu-dpk.net/?page=Radio E-mail : et...@manu-dpk.net MSN Messenger : m...@manu-dpk.net SKYPE : manu-dpk _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
Martin, This Grandstream HT503 makes it possible for me to send incoming PSTN-calls to an Asterisk-server on the local network (for IVR) ??? How about voice quality ? (the negative point of Linksys SPA) Best regards, Jonas. On Thu, 2009-11-12 at 18:59 -0500, Martin wrote: Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port has its own sip account. Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP traffic through Asterisk??
I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? Thank you very much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] little boy on asterisk and Debian
On Fri, Nov 13, 2009 at 11:31 AM, Manu et...@manu-dpk.net wrote: Can you help me please? Thank you very much. Voici un meilleur site pour poser des questions de tout genre en français : http://asterisk-france.net/ /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
Tilghman Lesher tles...@digium.com writes: Many consumer-grade switches effectively turn into hubs when more than 1023 MAC addresses are seen on a network. This may be done intentionally by somebody attempting to eavesdrop on all network connections sent through the switch. A reboot of the switch might (temporarily) remedy the problem, but you'd be better off getting an enterprise-grade switch that does not exhibit such misbehavior. Even so, all network cards automatically drop all unicast traffic not destined to their mac address (or addresses). This is turned off when the nic is in promiscuous mode, but that shouldn't happen on hardphones. Also, it is highly unlikely that IP stack wouldn't drop the traffic itself. This is simply too basic to get wrong. The only way I can see that a low-layer network problem could cause crosstalk is if two phones somehow acquired the same MAC address. They would likely end up with the same IP as well, and that could certainly cause problems. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POTS 4K linear codec
Cary Fitch ca...@usawide.net writes: Is there a plain 64K codec that would simply pass through the SIP server and be handed off to a PRI or phone co. trunk on a T1 on the other side of the SIP server? Digital 64K telco sounds very good as a phone conversation. You can't get a guaranteed bit-for-bit identical stream through SIP/RTP or IAX. You can pick the same codecs as the PSTN uses (Alaw or ulaw, depending on country), but jitter and packet loss still makes things like DTMF or fax/modem unreliable. For DTMF it is better to signal that in RTP or SIP, for fax you want T.38, and for modems you need incense and strange rites at midnight. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk systems hang with hfcmulti_rx no memory for rx_skb
Hi, I have two Asterisk systems that hang once every 4-6 days (more or less). One has * 1.2.31.1 and the other 1.4.26.2. The last system collapsed today and I saw several messages looping endlessly on screen: hfcmulti_rx no memory for rx_skb alloc_stack_skb(303,110): no skb size During this time the server is completely down (no ping). A forced reboot is necessary (very ugly situation in production environment as you can imagine). If hfcmulti_rx is the culprit then it should have something to do with receiving calls via ISDN lines. no memory suggests just that but my systems have 4GB RAM each and there aren't so many concurrent calls (max 18 ISDN channels). Both asterisk systems have a B410P 4-port ISDN card (http://www.digium.com/en/products/digital/b410p.php). # uname -a Linux voip1 2.6.23-gentoo-r8 #1 SMP Fri Oct 17 01:10:38 CEST 2008 i686 Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz GenuineIntel GNU/Linux misdn 1.1.7.2 misdnuser 1.1.7.2 Has anyone ever witnessed this issue with ISDN cards and Asterisk? What does Digium recommend for the B410P? (kernel version, misdn version, asterisk version) In other words, what should I try? (new kernel, new misdn?) Or should I move to DAHDI? But does DAHDI fully support B410P hardware? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
I basically agree, but I couldn't resist: On Fri, Nov 13, 2009 at 09:51:59AM +0100, Leif Neland wrote: Why should my call (and my money) go from my desk via my ip-pabc to my voisp possibly through pstn (through echelon) to your voisp to your ip-pabc to your desk, when it could go from my ip-pabc to your ip-pabc directly. Wow, can I get money from incoming calls? :-) (Money trasfers over IP) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] solution for NAT issues?
Does the phone have some sort of NAT Keepalive setting? Often, the only way to keep that port open on the user's NAT gateway is to have the NATted client send the occasional data out through the port. N. Ron wrote: i have also tried setting qualify='yes' but cpu usage spiked to 100%. Ron wrote: Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still registered on port 49923 but when the request is sent to that port the NAT router does not forward port 49923 to port of the IP phone, maybe nat mapping has expired or something. I have tried STUN, still sometimes the phones just cannot be reached. is there any other software to manage binding of ports on specific users so that the routers always keeps the port mapped to port of the ip phone . TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] solution for NAT issues?
hi sir, yes i am using Linksys SPA's i set NAT Mapping enable and NAT Keep-live to Yes. still sometimes the phone cannot be reach even though it is registered. regards ron SIP wrote: Does the phone have some sort of NAT Keepalive setting? Often, the only way to keep that port open on the user's NAT gateway is to have the NATted client send the occasional data out through the port. N. Ron wrote: i have also tried setting qualify='yes' but cpu usage spiked to 100%. Ron wrote: Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still registered on port 49923 but when the request is sent to that port the NAT router does not forward port 49923 to port of the IP phone, maybe nat mapping has expired or something. I have tried STUN, still sometimes the phones just cannot be reached. is there any other software to manage binding of ports on specific users so that the routers always keeps the port mapped to port of the ip phone . TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP traffic through Asterisk??
On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? snip I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: hi Dan
Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -Original Message- From: aster...@opensourcesolution.in [mailto:aster...@opensourcesolution.in] Sent: 13 November 2009 09:18 To: Dan Journo Subject: hi Dan Hi dan, sorry for sending u personal mail. i am a beginner in asterisk, i had configured a minimum dial plan in which i had made two extentions n made call between two extentions via soft phone (X-Lite). now i am begining with CALLER -ID, MUSIC ON HOLD, QUEUE.plz if u have any good link or documentation than share it with me. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
On Fri, 13 Nov 2009, aster...@opensourcesolution.in wrote: i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE GUYS UR HELP N SUPPORT WILL BE HIGHLY APPRECIATED. THX You are amazing. You could spend your time the reading many references you have been supplied yet you prefer to beg for others to do your job for you. Please go away. UR ABSENCE WILL BE HIGHLY APPRECIATED. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
This can also be caused by IRQ conflicts. You could try a different slot to see if it clears it up -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, November 12, 2009 1:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] my kernel is dazed and confused On Thu, Nov 12, 2009 at 09:31:11AM -0500, Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue NMI - Non Maskable Interrupt. This is a rather generic error message. Search a bit to see how to make some more sense of the messages following it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Its not just you mate. He's doing it to everyone, and sadly the list server is too clever to accept forged unsubscribes.. Steve On 13 Nov 2009, at 15:22, Dan Journo wrote: Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -Original Message- From: aster...@opensourcesolution.in [mailto:aster...@opensourcesolution.in ] Sent: 13 November 2009 09:18 To: Dan Journo Subject: hi Dan Hi dan, sorry for sending u personal mail. i am a beginner in asterisk, i had configured a minimum dial plan in which i had made two extentions n made call between two extentions via soft phone (X- Lite). now i am begining with CALLER -ID, MUSIC ON HOLD, QUEUE.plz if u have any good link or documentation than share it with me. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Sorry, I can't resist. How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. How about leaving member chastisement to the sponsor of the list? Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Be charitable and kind. Have a nice day. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, November 13, 2009 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan Its not just you mate. He's doing it to everyone, and sadly the list server is too clever to accept forged unsubscribes.. Steve On 13 Nov 2009, at 15:22, Dan Journo wrote: Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -Original Message- From: aster...@opensourcesolution.in [mailto:aster...@opensourcesolution.in ] Sent: 13 November 2009 09:18 To: Dan Journo Subject: hi Dan Hi dan, sorry for sending u personal mail. i am a beginner in asterisk, i had configured a minimum dial plan in which i had made two extentions n made call between two extentions via soft phone (X- Lite). now i am begining with CALLER -ID, MUSIC ON HOLD, QUEUE.plz if u have any good link or documentation than share it with me. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
This is happening here also :( On Fri, Nov 13, 2009 at 9:02 PM, Cary Fitch ca...@usawide.net wrote: Sorry, I can't resist. How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. How about leaving member chastisement to the sponsor of the list? Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Be charitable and kind. Have a nice day. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, November 13, 2009 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan Its not just you mate. He's doing it to everyone, and sadly the list server is too clever to accept forged unsubscribes.. Steve On 13 Nov 2009, at 15:22, Dan Journo wrote: Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -Original Message- From: aster...@opensourcesolution.in [mailto:aster...@opensourcesolution.in ] Sent: 13 November 2009 09:18 To: Dan Journo Subject: hi Dan Hi dan, sorry for sending u personal mail. i am a beginner in asterisk, i had configured a minimum dial plan in which i had made two extentions n made call between two extentions via soft phone (X- Lite). now i am begining with CALLER -ID, MUSIC ON HOLD, QUEUE.plz if u have any good link or documentation than share it with me. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
On 13 Nov 2009, at 16:02, Cary Fitch wrote: Sorry, I can't resist. Evidently How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. You just joined. How about leaving member chastisement to the sponsor of the list? I'd happily do so if it happened. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Well, most of the people on the list managed to do this without harassing people.. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Yes, and they shouldn't. Its called etiquette. Be charitable and kind. Have a nice day. I always am. Here, have a free hug, *hug* I find that helps when I'm feeling grumpy. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VUC to...@12 ET: Allison Smith
If you missed @voicegal last time or didn't go to Astricon, join us today on the Voip Users Conference to meet Allison Smith, the voice of Asterisk. Or go listen to the FBI talk about security... http://VoipUsersConference.org for details. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] destroy zombie session
What does the zombie call look like in core show channels? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aggio Alberto Sent: Friday, November 13, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] destroy zombie session Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that soft hangup should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command iax2 show channels) IP-AM-PBX*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)10.229.47.113REMOTE_SER 06818/14174 2/2 0ms -0001ms ms unknow 1 active IAX channel IP-AM-PBX*CLI Here I can't issue soft hangup command because I haven't a channel to specify (None is not a choice :-) ). Now the question is: is there a way to drop this (zombie) channel off and release frozen resources? Restarting asterisk is not an option (or maybe the last chance if I have no other way to achieve this result :-)) Thanks in advance for your replies. Cheers, Alberto Aggio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] destroy zombie session
Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that soft hangup should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command iax2 show channels) IP-AM-PBX*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)10.229.47.113REMOTE_SER 06818/14174 2/2 0ms -0001ms ms unknow 1 active IAX channel IP-AM-PBX*CLI Here I can't issue soft hangup command because I haven't a channel to specify (None is not a choice :) ). Now the question is: is there a way to drop this (zombie) channel off and release frozen resources? Restarting asterisk is not an option (or maybe the last chance if I have no other way to achieve this result :)) Thanks in advance for your replies. Cheers, Alberto Aggio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Sorry, I can't resist. How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. How about leaving member chastisement to the sponsor of the list? That's unlikely to happen in most cases. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Yes. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Be charitable and kind. Have a nice day. There's absolutely something to be said for that. On the other hand, there is also something to be said for making people exhaust the available resources prior to solving their problems for them. You can even be charitable and kind while doing so... Back in the '90's, I knew a really bright guy who knew Windows and Novell inside and out. He was just learning UNIXy stuff (FreeBSD in particular) and he was discovering that there was a lot of application for the stuff. He would frequently approach me, desperately seeking an answer to some general problem of some sort. I would typically give relatively vague answers, ending up essentially with a figure it out yourself. This frustrated him to no end, but he would do so. Later, he would come to me, almost always with a workable solution, at which point we would often discuss the ins and outs of several different options. His solution wasn't always the *best*, but it would always serve as a foundation for the rest. Years later, he thanked me. At the time, he didn't really appreciate what I was doing and didn't see the bigger picture. Looking back on it, I think he saw that I had always tried to aim him in a sensible direction before shoving him off on his own to figure it out. He eventually grew confident enough and capable enough that he would no longer need to ask for help. I can fix your problems for you, or I can teach you to be self- sufficient... which one is doing you more of a favor? It may seem more charitable and kind to simply give someone answers, but I do not think it actually is, at least in this sort of situation. As for the original poster? It's my impression, reading in between the lines, that he probably hasn't tried that hard. Asterisk on Linux is pretty straightforward, and MOH is probably not that rough to get running. On FreeBSD? That's a different thing. Bleh. But it's still better to do it on-list rather than selecting someone at random to go and bother. I don't think anyone will prevent you from being charitable and kind by providing answers to the guy's questions on the list though. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
As is occasionally pointed out by discerning people, questions are not dumb because they are posed by newbies, or because they reflect a lack of familiarity or knowledge of the product. Questions are dumb when they are formulated in a manner consistent with extreme intellectual laziness, expressive laziness and lack of consideration for the charity of one's audience at best, and brain damage at worst. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?
On Thu, 2009-11-12 at 20:18 -0700, Joseph wrote: Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. Compete, With which iax-ata ??? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
How do you know a good soccer player when you see one? If you are a good scout, just by his body language. Just by seeing him how he walks and positions himself on a field. By the time he touches the ball, he is either eliminated from my list of prospects or he is marked as good to be considered. How do you know a good technical person when you see one? Because irrespective of wherever he/she is from, regardless of his language, social status, and even upbringing, he KNOWS WHAT GOOGLE IS. The desire to investigate, research, and READ must be born with him the same day he is born. There is a saying that goes you can't tech a man anything, you can only help him find himself. In this case, this man has been attentively helped, and his questions have been duly answered more than once in this forum. He has been told in no uncertain terms to SEARCH IN GOOGLE... That is the answer, based in the fact that he will find plenty of solutions to HIS particular question within 1 minute. He doesn't listen. And listening is not a cultural constrain (perhaps if we call him little grasshopper will he be more attentive?). To me, he doesn't have it. I have friends who ask me all the time to teach them programming and I ignore them. Inside I know that they don't have it because a real programmer never asks another to teach him to program, therefore I would save myself the time. But maybe I am wrong, maybe this guy will learn from what we are trying to tell him... I hope. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, November 13, 2009 8:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan On 13 Nov 2009, at 16:02, Cary Fitch wrote: Sorry, I can't resist. Evidently How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. You just joined. How about leaving member chastisement to the sponsor of the list? I'd happily do so if it happened. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Well, most of the people on the list managed to do this without harassing people.. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Yes, and they shouldn't. Its called etiquette. Be charitable and kind. Have a nice day. I always am. Here, have a free hug, *hug* I find that helps when I'm feeling grumpy. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface
On Thu, 2009-11-12 at 18:59 -0500, Martin wrote: Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port has its own sip account. Martin - Original Message - From: jonas kellens To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 5:38 AM Subject: Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface I've read (through google) that the Linksys SPA-products do not have good voice quality on the PSTN-line. Grandstream HT486 is also just lifeline and EOL. The only I come up with is Patton-gateways but these are not at all cheap ! Jonas. On Thu, 2009-11-12 at 10:13 +, Steve Howes wrote: On 12 Nov 2009, at 09:33, jonas kellens wrote: I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the FXO-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my private Asterisk-server. Could someone advice on a gateway that can take analogue calls and transfer them on my local network ?! I know about the Digium-cards. Are there alternatives ? Google could tell you this Try the Linksys/Sipura type products S Digium iaxy are perhaps a bit flaky. But no problems (either stability or voice quality) with spa-3102, specially thoses since cisco took over... hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No dahdi_zttools in AsteriskNow?
Just picked up Asterisk the Future of Telephony, every other listed program is there (Book does not tell you about the changeover to dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools installed, tried to install via yum and says it is already installed. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No dahdi_zttools in AsteriskNow?
Just picked up Asterisk the Future of Telephony, every other listed program is there (Book does not tell you about the changeover to dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools installed, tried to install via yum and says it is already installed. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
I received this with a Sangoma card and CentOS 5.4. Downgrading to 5.2 resolved the issue. -- James Texter III Sr. Software Engineer NOBLE SYSTEMS 4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452 (o) 404.851.1331 ext. 357 (f) 404.851.1421 (e) jtext...@noblesys.com (w) www.noblesys.com We succeed when we exceed our customers expectations! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francesco Peeters Sent: Thursday, November 12, 2009 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] my kernel is dazed and confused Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike Googling for the error seems to indicate a possible kernel bug... Are you using Ubuntu or Debian?... -- Francesco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP source address error -- fixed
Quoting Jaap Winius jwin...@umrk.to: The question remains: how can a remote Asterisk server be receiving SIP packets that still contain the private net IP address of a client? Okay, I fixed it: by installing siproxd on the firewall system of the local network. With the Debian systems I'm running, I let iptables take care of NAT. Last December, with kernel 2.6.24, I didn't need a SIP proxy to get a SIP client to register with a remote Asterisk server. Now, with 2.6.26, I do. Conclusion: NAT sucks. If we were all using IPv6, this would not be an issue. Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No dahdi_zttools in AsteriskNow?
Humanx2000 schrieb: Just picked up Asterisk the Future of Telephony, every other listed program is there (Book does not tell you about the changeover to dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools installed, tried to install via yum and says it is already installed. zttool is now called dahdi_tool. dahdi_tabtab Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Joe Greco wrote: Sorry, I can't resist. How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. How about leaving member chastisement to the sponsor of the list? That's unlikely to happen in most cases. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Yes. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Be charitable and kind. Have a nice day. There's absolutely something to be said for that. On the other hand, there is also something to be said for making people exhaust the available resources prior to solving their problems for them. You can even be charitable and kind while doing so... Back in the '90's, I knew a really bright guy who knew Windows and Novell inside and out. He was just learning UNIXy stuff (FreeBSD in particular) and he was discovering that there was a lot of application for the stuff. He would frequently approach me, desperately seeking an answer to some general problem of some sort. I would typically give relatively vague answers, ending up essentially with a figure it out yourself. This frustrated him to no end, but he would do so. Later, he would come to me, almost always with a workable solution, at which point we would often discuss the ins and outs of several different options. His solution wasn't always the *best*, but it would always serve as a foundation for the rest. Years later, he thanked me. At the time, he didn't really appreciate what I was doing and didn't see the bigger picture. Looking back on it, I think he saw that I had always tried to aim him in a sensible direction before shoving him off on his own to figure it out. He eventually grew confident enough and capable enough that he would no longer need to ask for help. I can fix your problems for you, or I can teach you to be self- sufficient... which one is doing you more of a favor? It may seem more charitable and kind to simply give someone answers, but I do not think it actually is, at least in this sort of situation. As for the original poster? It's my impression, reading in between the lines, that he probably hasn't tried that hard. Asterisk on Linux is pretty straightforward, and MOH is probably not that rough to get running. On FreeBSD? That's a different thing. Bleh. But it's still better to do it on-list rather than selecting someone at random to go and bother. I don't think anyone will prevent you from being charitable and kind by providing answers to the guy's questions on the list though. ... JG Slightly paraphrasing a very old and wise saying: Give a man a fish, he eats for a day. Teach him how to fish, he eats for a lifetime. -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] openSuse 11.2 and dahdi-linux
OK, I know it's only just out today but this is what I get when compiling dahdi-linux. make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make -C /lib/modules/2.6.31.5-0.1-default/build SUBDIRS=/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.2.0.2/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default' make -C ../../../linux-2.6.31.5-0.1 O=/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default/. modules CC [M] /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.o /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c: In function ‘wctc4xxp_net_register’: /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:776: error: ‘struct net_device’ has no member named ‘set_multicast_list’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:777: error: ‘struct net_device’ has no member named ‘open’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:778: error: ‘struct net_device’ has no member named ‘stop’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:779: error: ‘struct net_device’ has no member named ‘hard_start_xmit’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:780: error: ‘struct net_device’ has no member named ‘get_stats’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:781: error: ‘struct net_device’ has no member named ‘do_ioctl’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c: In function ‘wctc4xxp_init_one’: /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:3460: warning: ‘DMA_nnBIT_MASK’ is deprecated make[5]: *** [/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.o] Error 1 make[4]: *** [/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp] Error 2 make[3]: *** [_module_/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi] Error 2 make[2]: *** [sub-make] Error 2 make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default' make: *** [modules] Error 2 But actually I only want dahdi dummy at the moment where could I modify the Makefile to just do this? Dave Cotton smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Home line noise problem
At 03:53 PM 11/12/2009, you wrote: I Have a home line connected to a tdm400p with 3 extensions and a siemens sip-dect , it seems to work fine but during a call there is always a digital squeal every so often does anyone know what this could be? If it sounds like the tones you get when you press a button on the phone it's something thinking a key was pressed and turning it into a proper tone. It happens more with some peoples voices than others. No clue how to stop it. If it's not that, I've no idea. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!
On 11/11/2009 02:08 PM, Ex Vito wrote: We've been experiencing some tough time regarding a new Asterisk installation connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional VPMADT032 echo cancellation module. It appears there may be a regression in dahdi-linux 2.2.0 with regards to the wcte12xp driver and the VPMADT032 module (as discussed https://issues.asterisk.org/view.php?id=15724). Would you be willing to try at least revision 7584 of http://svn.asterisk.org/svn/dahdi/linux/branches/2.2 and report your results on that issue? Extra information / question: A quick peek at https://issues.asterisk.org/view.php?id=15498nbn=18 also lead me to test loading the wcte12xp driver with vpmsupport=0. The system load behaviour is exactly the same. The idle load you're seeing can be a little misleading, but essentially, once you load the drivers for both the wctdm24xxp and wcte12xp, there is a fixed cost associated with continuously moving the TDM data to/from the card. The load imposed by the drivers would only go up after this point if a) software echocan is enabled, or b) you're conferencing many calls in the kernel. Otherwiseit's fixed. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Slightly paraphrasing a very old and wise saying: Give a man a fish, he eats for a day. Teach him how to fish, he eats for a lifetime. -- JohnM I see no teaching, just no help. He doesn't eat today or tomorrow either. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openSuse 11.2 and dahdi-linux
On 11/13/2009 01:11 PM, Dave Cotton wrote: OK, I know it's only just out today but this is what I get when compiling dahdi-linux. make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make -C /lib/modules/2.6.31.5-0.1-default/build SUBDIRS=/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.2.0.2/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default' make -C ../../../linux-2.6.31.5-0.1 O=/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default/. modules CC [M] /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.o /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c: In function ‘wctc4xxp_net_register’: /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:776: error: ‘struct net_device’ has no member named ‘set_multicast_list’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:777: error: ‘struct net_device’ has no member named ‘open’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:778: error: ‘struct net_device’ has no member named ‘stop’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:779: error: ‘struct net_device’ has no member named ‘hard_start_xmit’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:780: error: ‘struct net_device’ has no member named ‘get_stats’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:781: error: ‘struct net_device’ has no member named ‘do_ioctl’ /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c: In function ‘wctc4xxp_init_one’: /usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.c:3460: warning: ‘DMA_nnBIT_MASK’ is deprecated make[5]: *** [/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp/base.o] Error 1 make[4]: *** [/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctc4xxp] Error 2 make[3]: *** [_module_/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi] Error 2 make[2]: *** [sub-make] Error 2 make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default' make: *** [modules] Error 2 But actually I only want dahdi dummy at the moment where could I modify the Makefile to just do this? The easiest thing to do is comment out the following line in drivers/dahdi/Kbuild. obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_WCTC4XXP) += wctc4xxp/ Or you can grab the head of the 2.2 branch or trunk which has all the build issues for recent kernels resolved. Hopefully soon there will be a 2.2.1 release but it takes some time to run through the regression tests. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
On Fri, Nov 13, 2009 at 1:06 PM, John Millican jmilli...@sentinelcommunications.com wrote: Slightly paraphrasing a very old and wise saying: Give a man a fish, he eats for a day. Teach him how to fish, he eats for a lifetime. I've always liked... Build a man a fire stays warm for a day Catch a man on fire stays warm the rest of his life. Either way, I agree. -jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
What say you to the proposal that some approaches to seeking help are so ridiculous they should not be tolerated? Community standards neither conceive nor enforce themselves. -- Sent from mobile device On Nov 13, 2009, at 2:32 PM, Cary Fitch ca...@usawide.net wrote: Slightly paraphrasing a very old and wise saying: Give a man a fish, he eats for a day. Teach him how to fish, he eats for a lifetime. -- JohnM I see no teaching, just no help. He doesn't eat today or tomorrow either. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk SIP hangup
Hello all, How can I ask Asterisk to ignore a sip hang-up request for XX seconds from the beginning of the session? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
On Fri, 13 Nov 2009, Jon Moore wrote: I've always liked... Build a man a fire stays warm for a day Catch a man on fire stays warm the rest of his life. Funniest thing I've read all day :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi Tenant Asterisk Server ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkr9ucAACgkQvp6arS3vDioyfgCgimKexiFzTRnajuZmljDgHWEQ t9UAnidkNJd8r9hKsiEU4no9jglG7uNF =YHUR -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
I added some examples a while back to the extensions.conf.sample and the voicemail.conf.sample code to show how to support distinct domains for voice mail contexts... which was a big obstacle to multi-tenancy... otherwise, you couldn't have individual greetings, etc. For places (like Montreal and Bruxelles) where you need to further tailor context on a per-language basis, that's not been fully exercised... or for states like Indiana and Idaho that exist in two timezones... Actually, that's not entirely true. I tested having a default timezone and then overriding it on a per-SIP context basis and it seemed to work: https://issues.asterisk.org/view.php?id=16090 For POTS or ISDN this would be a little more work, but not impossible. See the [acme] stuff in extensions.conf.sample and voicemail.conf.sample and reply back (on list) if you have questions. -Philip On 11/13/2009 11:55 AM, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkr9ucAACgkQvp6arS3vDioyfgCgimKexiFzTRnajuZmljDgHWEQ t9UAnidkNJd8r9hKsiEU4no9jglG7uNF =YHUR -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openSuse 11.2 and dahdi-linux
On 13/11/09 20:42, Shaun Ruffell wrote: The easiest thing to do is comment out the following line in drivers/dahdi/Kbuild. obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_WCTC4XXP) += wctc4xxp/ Or you can grab the head of the 2.2 branch or trunk which has all the build issues for recent kernels resolved. Hopefully soon there will be a 2.2.1 release but it takes some time to run through the regression tests. Ok thanks that got it compiled but then this appeared Loading DAHDI hardware modules: WARNING: All config files need .conf: /etc/modprobe.d/dahdi.blacklist, it will be ignored in a future release. WARNING: All config files need .conf: /etc/modprobe.d/options, it will be ignored in a future release. WARNING: All config files need .conf: /etc/modprobe.d/dahdi, it will be ignored in a future release. The files need an .conf extension now. Still, biggest problem is asterisk does not get any further than *CLI == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 I look farther tomorrow. Dave Cotton smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
snip What say you to the proposal that some approaches to seeking help are so ridiculous they should not be tolerated? /snip I say give me a break. Pre-judging people doesn't work on mailing lists given the inherent language barriers, etc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
David Gibbons wrote: snip What say you to the proposal that some approaches to seeking help are so ridiculous they should not be tolerated? /snip I say give me a break. Pre-judging people doesn't work on mailing lists given the inherent language barriers, etc. It can. There are intelligent and respectful ways of asking a question - as well as good ways to make the choices surrounding the asking of the question (such as emailing a dozen people privately and continuously badgering them, while posting to the list in parallel) - despite the encumbrance language differences pose. I've seen people with far, far worse English than our opensourcesolutions friend interact in a way that reflects much more common sense, awareness, respect of the audience, and a generally cultured way of seeking assistance that dignifies a response. -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
What say you to the proposal that some approaches to seeking help are so ridiculous they should not be tolerated? Community standards neither conceive nor enforce themselves. This community standard is entirely self-enforcing. If everybody thinks the request for help is unwarranted and doesn't deserve an answer, then nobody answers. If it isn't so intolerable to fall victim to that, then someone may feel inclined to help out and answer. If the volume of such requests becomes sufficiently burdensome that it exhausts those answering the questions, then eventually the equilibrium resets; those answering the questions begin to pick out the ones that they feel are worthy of answers. Years ago, I got a bit of a panicky phone call from a sysadmin at an ISP that I was loosely familiar with from mailing lists. He was rather frazzled and puzzled because he had been struggling to solve a problem during a downtime; it was something I was familiar with and had been advocating on a mailing list. He was doing something completely reasonable-seeming, it's just that what he was doing didn't work, and had never worked that way. I walked him through a different method, from memory, solved his problem. I'm positive I could have charged him billable hours, but I didn't, because I felt somewhat responsible for having advocated something rather complex that was followed by a competent admin and it blew up in his face - precisely *because* the problem and fix were obscure. The Asterisk community is great at promoting itself, but quite frankly the documentation and solutions are sometimes not all that great. It can be challenging to find the right fix, or even *a* fix. Questions *must* be expected. The community has generally been fairly successful at coping with the questions; I view the beginning of this thread to be a sign of that. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
On Fri, 13 Nov 2009, Cary Fitch wrote: Sorry, I can't resist. Next time please try harder. How do I join the Mail List Nazi Corp? And of course, lacking any sense of history, let's blame it on the Nazis. Asking neophyte questions are objected to by some, top posting by those who blast others, etc. Not at all. If you had any sense of Pawan's history you may have chosen sides differently: Date: Tue, 27 Oct 2009 09:34:18 + Subject: [asterisk-users] Installing Asterisk Pawan states he is reading an Asterisk book and requests suggestions on which OS he should use. He received helpful responses from Dan Journo, PATRICK KANGETHE, John Novack, and Hans Witvliet. 30 minutes later he posts a brilliant tome Subjected installing consisting of 2 words -- installing asterisk. He received less than helpful responses from Steve Howes, Alex Balashov, and Pascal Bruno. Date: Wed, 28 Oct 2009 14:07:30 + Subject: [asterisk-users] deploying asterisk Pawan states he had just finished the installation requirement of asterisk and now feels competent to piss off 40 executives with his first installation. He received helpful responses from Danny Nicholas, Darrick Hartman, Steve Edwards (me), and Alex Balashov. Date: Mon, 02 Nov 2009 09:37:42 + Subject: [asterisk-users] hardware requirements for asterisk Pawan request help with hardware requirements. Curiously, he implies that he can read and has just finished my chapters of asterisk. He receives helpful responses from Alex Balashov and Hans Witvliet. Date: Fri, 06 Nov 2009 04:33:09 + Subject: [asterisk-users] asterisk,libpri,zaptel Pawan requests help installing Asterisk. Date: Fri, 06 Nov 2009 17:08:02 + Subject: [asterisk-users] problem while compiling asterisk tar file Pawan requests help in compiling gtk. He receives helpful responses from Jimmy Godbout, Danny Nicholas, Steve Howes, Jason Parker. Date: Sat, 07 Nov 2009 17:29:57 + Subject: [asterisk-users] help in installing asterisk Pawan requests help in compiling Asterisk. Date: Sun, 08 Nov 2009 06:20:46 + Subject: [asterisk-users] how to check version of asterisk Pawan requests help to determine the version of Asterisk he installed. He receives helpful responses from Alex Balashov, Tzafrir Cohen, and C. Savinovich. Date: Mon, 09 Nov 2009 17:11:47 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan requests help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Matt Riddell and Danny Nicholas. He receives less that helpful responses from Alex Balashov, Steve Howes, and C. Savinovich in response to emailing them privately. Date: Tue, 10 Nov 2009 18:16:50 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan solicits help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Alex Balashov and Barry L. Kline. Date: Thu, 12 Nov 2009 06:31:35 + Subject: [asterisk-users] soft phone (X-lite) not able to register with asterisk Pawan solicits help configuring a softphone. 20 minutes later he posts the same request. He receives a helpful response from ABBAS SHAKEEL. Date: Fri, 13 Nov 2009 08:47:41 + Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE Pawan indicates he has succeeded in placing a call between 2 extensions and now wants someone to complete is dialplan. He receives a helpful response from Leif Neland and less than helpful responses from Steve Howes and Steve Edwards. He also invites a flame-fest by soliciting help privately from several list members. All this in the last 2 weeks. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Distance is no defense to ignorance. If you have the ability to email, you have access to all the resources you need -- if given a gentle nudge or 2. Pawan has demonstrated a lack of social and technical skills that a clue-by-4 can't cure. Time to move on... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
I strongly concur with this realistic and very well-researched thesis. -- Sent from mobile device On Nov 13, 2009, at 5:59 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 13 Nov 2009, Cary Fitch wrote: Sorry, I can't resist. Next time please try harder. How do I join the Mail List Nazi Corp? And of course, lacking any sense of history, let's blame it on the Nazis. Asking neophyte questions are objected to by some, top posting by those who blast others, etc. Not at all. If you had any sense of Pawan's history you may have chosen sides differently: Date: Tue, 27 Oct 2009 09:34:18 + Subject: [asterisk-users] Installing Asterisk Pawan states he is reading an Asterisk book and requests suggestions on which OS he should use. He received helpful responses from Dan Journo, PATRICK KANGETHE, John Novack, and Hans Witvliet. 30 minutes later he posts a brilliant tome Subjected installing consisting of 2 words -- installing asterisk. He received less than helpful responses from Steve Howes, Alex Balashov, and Pascal Bruno. Date: Wed, 28 Oct 2009 14:07:30 + Subject: [asterisk-users] deploying asterisk Pawan states he had just finished the installation requirement of asterisk and now feels competent to piss off 40 executives with his first installation. He received helpful responses from Danny Nicholas, Darrick Hartman, Steve Edwards (me), and Alex Balashov. Date: Mon, 02 Nov 2009 09:37:42 + Subject: [asterisk-users] hardware requirements for asterisk Pawan request help with hardware requirements. Curiously, he implies that he can read and has just finished my chapters of asterisk. He receives helpful responses from Alex Balashov and Hans Witvliet. Date: Fri, 06 Nov 2009 04:33:09 + Subject: [asterisk-users] asterisk,libpri,zaptel Pawan requests help installing Asterisk. Date: Fri, 06 Nov 2009 17:08:02 + Subject: [asterisk-users] problem while compiling asterisk tar file Pawan requests help in compiling gtk. He receives helpful responses from Jimmy Godbout, Danny Nicholas, Steve Howes, Jason Parker. Date: Sat, 07 Nov 2009 17:29:57 + Subject: [asterisk-users] help in installing asterisk Pawan requests help in compiling Asterisk. Date: Sun, 08 Nov 2009 06:20:46 + Subject: [asterisk-users] how to check version of asterisk Pawan requests help to determine the version of Asterisk he installed. He receives helpful responses from Alex Balashov, Tzafrir Cohen, and C. Savinovich. Date: Mon, 09 Nov 2009 17:11:47 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan requests help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Matt Riddell and Danny Nicholas. He receives less that helpful responses from Alex Balashov, Steve Howes, and C. Savinovich in response to emailing them privately. Date: Tue, 10 Nov 2009 18:16:50 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan solicits help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Alex Balashov and Barry L. Kline. Date: Thu, 12 Nov 2009 06:31:35 + Subject: [asterisk-users] soft phone (X-lite) not able to register with asterisk Pawan solicits help configuring a softphone. 20 minutes later he posts the same request. He receives a helpful response from ABBAS SHAKEEL. Date: Fri, 13 Nov 2009 08:47:41 + Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE Pawan indicates he has succeeded in placing a call between 2 extensions and now wants someone to complete is dialplan. He receives a helpful response from Leif Neland and less than helpful responses from Steve Howes and Steve Edwards. He also invites a flame-fest by soliciting help privately from several list members. All this in the last 2 weeks. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Distance is no defense to ignorance. If you have the ability to email, you have access to all the resources you need -- if given a gentle nudge or 2. Pawan has demonstrated a lack of social and technical skills that a clue-by-4 can't cure. Time to move on... -- Thanks in advance, --- -- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users
Re: [asterisk-users] FW: hi Dan
My point was the two previous posters could have ignored the request and made no post at all. That they were violating a rule by top posting to tell a person not to bug them. And, someone criticized me for an off topic post and of course there have been 15-20 more. And some have top posted and interleave posted, and etc. And, it will all die down in a day or so. It is Friday night, time to turn off the computer and click Mark all as read on Monday morning. Be charitable and kind. Have a nice weekend. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, November 13, 2009 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan On Fri, 13 Nov 2009, Cary Fitch wrote: Sorry, I can't resist. Next time please try harder. How do I join the Mail List Nazi Corp? And of course, lacking any sense of history, let's blame it on the Nazis. Asking neophyte questions are objected to by some, top posting by those who blast others, etc. Not at all. If you had any sense of Pawan's history you may have chosen sides differently: Date: Tue, 27 Oct 2009 09:34:18 + Subject: [asterisk-users] Installing Asterisk Pawan states he is reading an Asterisk book and requests suggestions on which OS he should use. He received helpful responses from Dan Journo, PATRICK KANGETHE, John Novack, and Hans Witvliet. 30 minutes later he posts a brilliant tome Subjected installing consisting of 2 words -- installing asterisk. He received less than helpful responses from Steve Howes, Alex Balashov, and Pascal Bruno. Date: Wed, 28 Oct 2009 14:07:30 + Subject: [asterisk-users] deploying asterisk Pawan states he had just finished the installation requirement of asterisk and now feels competent to piss off 40 executives with his first installation. He received helpful responses from Danny Nicholas, Darrick Hartman, Steve Edwards (me), and Alex Balashov. Date: Mon, 02 Nov 2009 09:37:42 + Subject: [asterisk-users] hardware requirements for asterisk Pawan request help with hardware requirements. Curiously, he implies that he can read and has just finished my chapters of asterisk. He receives helpful responses from Alex Balashov and Hans Witvliet. Date: Fri, 06 Nov 2009 04:33:09 + Subject: [asterisk-users] asterisk,libpri,zaptel Pawan requests help installing Asterisk. Date: Fri, 06 Nov 2009 17:08:02 + Subject: [asterisk-users] problem while compiling asterisk tar file Pawan requests help in compiling gtk. He receives helpful responses from Jimmy Godbout, Danny Nicholas, Steve Howes, Jason Parker. Date: Sat, 07 Nov 2009 17:29:57 + Subject: [asterisk-users] help in installing asterisk Pawan requests help in compiling Asterisk. Date: Sun, 08 Nov 2009 06:20:46 + Subject: [asterisk-users] how to check version of asterisk Pawan requests help to determine the version of Asterisk he installed. He receives helpful responses from Alex Balashov, Tzafrir Cohen, and C. Savinovich. Date: Mon, 09 Nov 2009 17:11:47 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan requests help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Matt Riddell and Danny Nicholas. He receives less that helpful responses from Alex Balashov, Steve Howes, and C. Savinovich in response to emailing them privately. Date: Tue, 10 Nov 2009 18:16:50 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan solicits help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Alex Balashov and Barry L. Kline. Date: Thu, 12 Nov 2009 06:31:35 + Subject: [asterisk-users] soft phone (X-lite) not able to register with asterisk Pawan solicits help configuring a softphone. 20 minutes later he posts the same request. He receives a helpful response from ABBAS SHAKEEL. Date: Fri, 13 Nov 2009 08:47:41 + Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE Pawan indicates he has succeeded in placing a call between 2 extensions and now wants someone to complete is dialplan. He receives a helpful response from Leif Neland and less than helpful responses from Steve Howes and Steve Edwards. He also invites a flame-fest by soliciting help privately from several list members. All this in the last 2 weeks. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Distance is no defense to ignorance. If you have the ability to email, you have access to all the resources you need --
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
On Nov 13, 2009, at 6:16 PM, Cary Fitch wrote: My point was the two previous posters could have ignored the request and made no post at all. That they were violating a rule by top posting to tell a person not to bug them. And, someone criticized me for an off topic post and of course there have been 15-20 more. And some have top posted and interleave posted, and etc. And, it will all die down in a day or so. Be charitable and kind. Have a nice weekend. Cary Fitch Just the other day, I was commenting how helpful people are on this list. We rarely get into flame wars. We rarely port RTFM. And yet, there comes a time when even the most flame-proof must flame. I think it's been shown that people have been BEYOND helpful, more than kind, very patient, and courteous... up until the point where it was time not to be. I just want to say thank you to those that continue to be helpful day in and out and say... I support their handling of the 100%. That being said... when we discuss our flames, we shall not invoke Godwin's Law. :) ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote: On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. snip Thanks for the update. How will it be available at that point? Will there be an immediate 1.6.1.9 release or will it only be via SVN? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Xorcom Astribank udev issue in Ubuntu 9.10
Hi, I have upgraded an Asterisk installation with a Xorcom BRI Astribank that was working under Ubuntu 8.04 to Ubuntu 9.10 and the device is no longer initialized. When I reload the udev rules, I see that the rules seems to be correctly loaded: udevd[452]: reading '/etc/udev/rules.d/40-xpp.rules' as rules file However, these rules do not seem to be triggered: # udevadm trigger --attr-match=idVendor=e4e4 --action add UDEV [1258155881.753357] add /devices/pci:00/:00:1a.7/usb1/1-1 (usb) UDEV_LOG=7 ACTION=add DEVPATH=/devices/pci:00/:00:1a.7/usb1/1-1 SUBSYSTEM=usb DEVNAME=/dev/bus/usb/001/009 DEVTYPE=usb_device DRIVER=usb DEVICE=/proc/bus/usb/001/009 PRODUCT=e4e4/1141/ TYPE=0/0/0 BUSNUM=001 DEVNUM=009 SEQNUM=1516 ID_VENDOR=Xorcom_LTD ID_VENDOR_ENC=Xorcom\x20LTD ID_VENDOR_ID=e4e4 ID_MODEL=Astribank ID_MODEL_ENC=Astribank ID_MODEL_ID=1141 ID_REVISION= ID_SERIAL=Xorcom_LTD_Astribank_ ID_SERIAL_SHORT= ID_BUS=usb ID_USB_INTERFACES=:ff: MAJOR=189 MINOR=8 DEVLINKS=/dev/char/189:8 Nov 14 00:44:41 asterisk udevd[452]: seq 1516 queued, 'add' 'usb' Nov 14 00:44:41 asterisk udevd[452]: passed 287 bytes to monitor 0xb94cf1f0 Nov 14 00:44:41 asterisk udevd-work[4995]: seq 1516 running Nov 14 00:44:41 asterisk udevd-work[4995]: device 0xb94d7700 has devpath '/devices/pci:00/:00:1a.7/usb1/1-1' Nov 14 00:44:41 asterisk udevd-work[4995]: IMPORT 'usb_id --export /devices/pci:00/:00:1a.7/usb1/1-1' /lib/udev/rules.d/40-libgphoto2-2.rules:11 Nov 14 00:44:41 asterisk udevd-work[4995]: 'usb_id --export /devices/pci:00/:00:1a.7/usb1/1-1' started Nov 14 00:44:41 asterisk usb_id[4997]: custom logging function 0xb80a5008 registered Nov 14 00:44:41 asterisk usb_id[4997]: device 0xb80a50e8 has devpath '/devices/pci:00/:00:1a.7/usb1/1-1' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_VENDOR=Xorcom_LTD' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_VENDOR_ENC=Xorcom\x20LTD' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_VENDOR_ID=e4e4' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_MODEL=Astribank' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_MODEL_ENC=Astribank' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_MODEL_ID=1141' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_REVISION=' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_SERIAL=Xorcom_LTD_Astribank_' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_SERIAL_SHORT=' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_BUS=usb' Nov 14 00:44:41 asterisk udevd-work[4995]: '/lib/udev/usb_id' (stdout) 'ID_USB_INTERFACES=:ff:' Nov 14 00:44:41 asterisk udevd-work[4995]: 'usb_id --export /devices/pci:00/:00:1a.7/usb1/1-1' returned with exitcode 0 Nov 14 00:44:41 asterisk udevd-work[4995]: device 0xb94decf8 has devpath '/devices/pci:00/:00:1a.7/usb1' Nov 14 00:44:41 asterisk udevd-work[4995]: device 0xb94df440 has devpath '/devices/pci:00/:00:1a.7' Nov 14 00:44:41 asterisk udevd-work[4995]: device 0xb94df5e0 has devpath '/devices/pci:00' Nov 14 00:44:41 asterisk udevd-work[4995]: LINK 'char/189:8' /lib/udev/rules.d/50-udev-default.rules:4 Nov 14 00:44:41 asterisk udevd-work[4995]: MODE 0664 /lib/udev/rules.d/50-udev-default.rules:58 Nov 14 00:44:41 asterisk udevd-work[4995]: RUN 'socket:@/org/freedesktop/hal/udev_event' /lib/udev/rules.d/90-hal.rules:2 Nov 14 00:44:41 asterisk udevd-work[4995]: no node name set, will use kernel supplied name 'bus/usb/001/009' Nov 14 00:44:41 asterisk udevd-work[4995]: created db file for '/devices/pci:00/:00:1a.7/usb1/1-1' in '/dev/.udev/db/usb:1-1' Nov 14 00:44:41 asterisk udevd-work[4995]: creating device node '/dev/bus/usb/001/009', devnum=189:8, mode=0664, uid=0, gid=0 Nov 14 00:44:41 asterisk udevd-work[4995]: mknod(/dev/bus/usb/001/009, 020664, (189,8)) Nov 14 00:44:41 asterisk udevd-work[4995]: chmod(/dev/bus/usb/001/009, 020664) Nov 14 00:44:41 asterisk udevd-work[4995]: chown(/dev/bus/usb/001/009, 0, 0) Nov 14 00:44:41 asterisk udevd-work[4995]: creating symlink '/dev/char/189:8' to '../bus/usb/001/009' Nov 14 00:44:41 asterisk udevd-work[4995]: passed 593 bytes to monitor 0xb94df948 Nov 14 00:44:41 asterisk udevd-work[4995]: passed -1 bytes to monitor 0xb94deb80 Nov 14 00:44:41 asterisk udevd-work[4995]: seq 1516 processed with 0 Nov 14 00:44:41 asterisk udevd[452]: seq 1516 done with 0 I have tried with the version of dahdi that comes with Ubuntu 9.10 and I am now trying with the svn version of dahdi with no more success. Does anyone have an idea how I can solve this issue? Thanks, Eric -- Eric van der Vlist v...@dyomedea.com Dyomedea (http://dyomedea.com)
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On 18:55, Fri 13 Nov 09, John A. Sullivan III wrote: On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote: On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. snip Thanks for the update. How will it be available at that point? Will there be an immediate 1.6.1.9 release or will it only be via SVN? - John not sure yet. Will have a look at it tomorrow and get back to you here ok ? -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:How to stop Asterisk?
Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod asterisk Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to stop Asterisk?
cli stop now or cli stop gracefully :) otherwise pkill -9 asterisk On Sat, Nov 14, 2009 at 7:39 AM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod asterisk Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Yawar Hadi Noshahi Consultant/Software Engineer NGI Islamabad MS Computer Science Linkoping University Sweden +46700-445479 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
