[asterisk-users] [CID] Unknown IE 18/21?
Hello Apparently, those are just warnings, but I'd like to know what those messages mean: [Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 18 [Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 21 Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specific SIP answers on incoming calls?
Stefan Gofferje wrote: Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. Wrong number to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides wrong number, I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to send a specific reply out of the dialplan? No. The dial plan does not provide such low-level access to the SIP stack. All that you can do is call the dial plan applications, and they will do whatever it is they do on the SIP level. Also, keep in mind that the abstraction you are suggesting is leaky. The cause codes that Hangup() can send on the ISDN side are still *disconnection*-related cause codes. You can't just send any kind of Q.931 / 921 response arbitrarily, even if it is valid from the perspective of the protocol's state machine. That is logically analogous to what you are asking to do with SIP here. Asterisk is not designed to expose low-level protocol details; it's a relatively high-level application. If you want to translate various SIP states Asterisk puts out into customised responses, you will probably need to use something like Kamailio / OpenSIPS (OpenSER) outboard with Asterisk to perform those translations. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
Hi Hitesh, Usually, subscribing to DID provider is a one way thing, they can call you to that number, but you cannot call out via that number. If you already have a pots line available, which means you are probably paying monthly for it already, might as well buy an fxo card and make use of the line. anyone in india can call you locally and you can call anyone in india using the same line, as everyone is suggesting, use trixbox, not much linux experience is required, just boot from the cd and let it install itself. you might need linux experience when you compile drivers for your fxo card though, but they usually come with instructions which is quite easy to follow. hth regards, nhadie logan wrote: Hi Jai, If I understand correctly then the DID will enable to call me on the hardphone connected to the Asterisk. Will it also enable me to call out using the PSTN line at my home in India from Canada? Thanks. Best REgards, Hitesh On Fri, Sep 19, 2008 at 10:33 AM, Jai Rangi [EMAIL PROTECTED] wrote: Hitesh, If you dont have experience with Linux I would recommend you to use Trixbox, that will come with all the required packages and will do everythign for you. Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can buy DIDs that can come to your asterisk over the internet. Jai www.didforsale.com *Buy SIP DIDs at low cost unlimited minutes http://www.didforsale.com; On Fri, Sep 19, 2008 at 9:18 AM, logan [EMAIL PROTECTED] wrote: Hello Ram, Thanks for the response. As I said there are too many options out there :). Could you help me in settling down on one? Something that will work with the phone lines in India is just fine for me. I don't have any or much Linux experience, but willing to play around, so any compatible distro will do for me. So once again: Which Linux distro is best with Asterisk? Which hardphone is the easiest to setup? Which fxo/fxs card I should go for? Thanks a lot guys. Best Regards, Hitesh On Thu, Sep 18, 2008 at 10:33 PM, ram [EMAIL PROTECTED] wrote: On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote: Thanks a lot Nhadie. I appreciate your help. Could you also suggest some brands or models of the FXO+FXS card that are seamlessly compatible to Asterisk? Also what hardphone I should go for as there are so many in the market? What should be the configuration of the system running this kind of Asterisk setup? And which Linux distribution is best suited with Asterisk? Hi you can look this compatable hardware http://www.voip-info.org/wiki/ http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems http://www.voip-info.org/wiki/view/VOIP+Phones Its very difficult to say which OS is good, its all depends on your experience and your hands on the same. Look at Trixbox, its automated CD ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone flying with BA on Monday for Astricon ?
Greetings to all ! I will be flying on Monday with BA from London to Phoenix so i was wondering if anyone else is on the same plane so there will more than inflight movies to pass the time :) If so please contact me off-list and we can arrange to meet. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.0-rc6 - SIP hold logic broken?
Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4 latest, everything worked as supposed. As this problem appears also between SIP clients, it is NOT a chan_sccp-related issue. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specific SIP answers on incoming calls?
Hi! specific SIP headers. Besides wrong number, I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to send a specific reply out of the dialplan? No. The dial plan does not provide such low-level access to the SIP stack. All that you can do is call the dial plan applications, and they will do whatever it is they do on the SIP level. You might want to try the app Transfer() to send a SIP REFER. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+transfer Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting results messages from CLI commands via -rx
On Fri, Sep 19, 2008 at 12:54:58PM -0700, George Williams wrote: Hi, I am issuing CLI commands via script, using the asterisk -rx method. Its working great. Now, I need to get the results of the command to look for error messages, etc. I've tried setting several -v flags as well, but I only get the Asterisk startup text (version, license info, etc), not the results of the command itself. Don't use v-s. They are global. It seems you didn't have the '-x'. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
On Fri, Sep 19, 2008 at 5:29 AM, [EMAIL PROTECTED] wrote: Hi, I have a SIP request which comes from an Asterisk and which has to re-enter in the same Asterisk (during the same session), but during the second passage in Asterisk, it send me a 482 Loop Detected. So is it a bug or Asterisk control the session and considere it as a loop ? If it is not a bug, how could I resolve this problem ? Try setting pedantic=yes in your sip.conf. -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco acquires Jabber
Wow - now this interesting http://www.techcrunch.com/2008/09/19/cisco-acquires-jabber-for-enterpris e-im/ I wonder what this means in the long run for the open development of this platform? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco acquires Jabber
I wonder what this means in the long run for the open development of this platform? Not a darn thing, unless Cisco screws around and makes an incompatible version of a jabber server and client that doesn't play according to the protocol. Microsoft Java, anybody? We'll see how long this list stays true: http://www.jabber.com/CE/JabberXCPInteroperabilityOptions Cisco didn't buy the protocol, and literally dozens of open-source projects that use the protocol in various ways are not affected by this. They bought one commercial implementation of a Jabber server (arguably multiple implementations). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callwaiting callerid
Is it possible to send callwaiting callerid to a channel without actually having a call waiting, I'm thinking if messaging a handset through the manger interface for example? Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco acquires Jabber
No I know they just bought the company and not the protocol basically they bought engineering bums on seats. http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.html Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. You could possible see different Cisco devices communicating with each other (or even using an api to communicate with other manufacturers devices) eg, you might have an XMPP api to 'discover' appliance functionality or to communicate status updates. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Saturday, 20 September 2008 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco acquires Jabber I wonder what this means in the long run for the open development of this platform? Not a darn thing, unless Cisco screws around and makes an incompatible version of a jabber server and client that doesn't play according to the protocol. Microsoft Java, anybody? We'll see how long this list stays true: http://www.jabber.com/CE/JabberXCPInteroperabilityOptions Cisco didn't buy the protocol, and literally dozens of open-source projects that use the protocol in various ways are not affected by this. They bought one commercial implementation of a Jabber server (arguably multiple implementations). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco acquires Jabber
On Sat, Sep 20, 2008 at 12:18:42PM -0400, Dean Collins wrote: No I know they just bought the company and not the protocol basically they bought engineering bums on seats. [1]http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.ht ml Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. You could possible see different Cisco devices communicating with each other (or even using an api to communicate with other manufacturers devices) eg, you might have an XMPP api to 'discover' appliance functionality or to communicate status updates. Jabber.com are in some big US gov departments, these are probably just the bodies Cisco want to get into with their UM systems. Making Cisco's UM based on Jabber and buying the expertise probably is a wise move for them. Also gives them interoperability with other systems ... Then move it down into the SME market as Linksys appliance. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
Moy, How i can do to join asterisk-r2 list ? My congratulations about your article in digium blog http://blogs.digium.com/page/2/ I will collaborate in your project and give support from Venezuela. Regards, Luis Morales On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva [EMAIL PROTECTED] wrote: Dae, If you can assist to my session may be we can discuss this issue you are having. I am about to add Colombia support for OpenR2, and even if you want to stick with Unicall I'd like to see what's going on there :-) Guys, just for your information, as of today, there is now an asterisk-r2 mailing list, where you may want to move this thread to (or not) :-) More and more I focus on Unicall and OpenR2 issues only, which I and other R2 users can monitor in asterisk-r2 more easily. Moy On Fri, Sep 19, 2008 at 1:10 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Yes I can see the channels with UC show channels, and it's says Idle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Friday, September 19, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Not really the unicall setup must be idem. So you can see the unicall channels ? It's moises are busy i can give you support too Regards, Luis Morales On Sat, Sep 20, 2008 at 12:15 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hi Luis, But this E1 has 30 channels, all for both directions... I must differentiate this?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Friday, September 19, 2008 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Hi dae, Your zapata.conf must be ok, now inyour unicall.conf [channels] language=es context=from-pstn usecallerid=yes hidecallerid=no immediate=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 loglevel=255 callgroup=1 pickupgroup=1 group=1 musiconhold=default relaxdtmf=yes category=NATIONAL_SUBSCRIBER ; protocolclass=mfcr2 protocolvariant=co,20,4,x,T1=1500,T2=24000,T3=15000,max-seize-wait-ack=3000 protocolend=cpe ; ; E1 IN group = 1 context = from-pstn channel = 1-15 ; E1 OUT group = 2 context = from-pstn channel = 17-31 - It's very important that you identify: - E1 lines in - E1 lines out Now from your cosole type: asterisk -r now from asterisk cli type: pbx* UC show channels The result must be an list with your unicall channels, similar to: Channel Extension Context Status Language MusicOnHold 1from-pstn Idle es default 2from-pstn Idle es default 3from-pstn Idle es default 4from-pstn Idle es default 5from-pstn Idle es default 6from-pstn Idle es default 7from-pstn Idle es default 8from-pstn Idle es default 9from-pstn Idle es default 10from-pstn Idle es default 11from-pstn Idle es default 12from-pstn Idle es default 13from-pstn Idle es default 14from-pstn Idle es default 15from-pstn Idle es default 17 6842 from-pstn Idle es default 18from-pstn Idle es default 19from-pstn Idle es default 20from-pstn Idle es default 21from-pstn Idle es default 22from-pstn Idle es default 23from-pstn Idle es default 24from-pstn Idle es default 25from-pstn Idle es default 26from-pstn Idle es default 27from-pstn Idle es default 28from-pstn Idle es default 29from-pstn Idle es default 30from-pstn Idle es default 31from-pstn Idle es default Good luck! Luis Morales On Fri, Sep 19, 2008 at 9:04 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: It's a Digium TE121P with Echo Cancellation Zapata.conf # Span 1: WCT1/0 Wildcard TE121 Card
Re: [asterisk-users] Cisco acquires Jabber
Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. Dean, I'm right there with you. My money is on them using it as the first step in a larger strategy to provide a framework for applications to run on network without needing an operating system. Think Amazon Elastic Cloud (with P2P and presence built in) but for applications. On Sat, Sep 20, 2008 at 11:18 AM, Dean Collins [EMAIL PROTECTED] wrote: No I know they just bought the company and not the protocol basically they bought engineering bums on seats. http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.html Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. You could possible see different Cisco devices communicating with each other (or even using an api to communicate with other manufacturers devices) eg, you might have an XMPP api to 'discover' appliance functionality or to communicate status updates. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.cognation.net/profile -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Saturday, 20 September 2008 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco acquires Jabber I wonder what this means in the long run for the open development of this platform? Not a darn thing, unless Cisco screws around and makes an incompatible version of a jabber server and client that doesn't play according to the protocol. Microsoft Java, anybody? We'll see how long this list stays true: http://www.jabber.com/CE/JabberXCPInteroperabilityOptions Cisco didn't buy the protocol, and literally dozens of open-source projects that use the protocol in various ways are not affected by this. They bought one commercial implementation of a Jabber server (arguably multiple implementations). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Weavver. Your voice, just better. Business Development: +1-714-726-8071 XMPP: mitchel.at.weavver.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] broadcast ability
I have asterisk 1.4 set up, using MySQL, with about 5,000 voicemail accounts. All calls to accounts are for leaving voicemail only, which is then emailed to the users. No voicemail is kept on the system, or in a db. I am interested in sending a pre-recorded broadcast message to all accounts. Can anyone provide me with some thoughts on how to do this as easily and efficiently as possible? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] broadcast ability
DesRae Mason wrote: accounts. Can anyone provide me with some thoughts on how to do this as easily and efficiently as possible? Generate a listing of email addresses from your database, record your broadcast message and email it to them? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
Luis, you can join asterisk-r2 mailing list in the same way you joined asterisk-users, just go to http://lists.digium.com/ and select asterisk-r2 mailing list, there you just need to provide your e-mail address. Moy On Sat, Sep 20, 2008 at 11:48 AM, Luis Morales [EMAIL PROTECTED] wrote: Moy, How i can do to join asterisk-r2 list ? My congratulations about your article in digium blog http://blogs.digium.com/page/2/ I will collaborate in your project and give support from Venezuela. Regards, Luis Morales On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva [EMAIL PROTECTED] wrote: Dae, If you can assist to my session may be we can discuss this issue you are having. I am about to add Colombia support for OpenR2, and even if you want to stick with Unicall I'd like to see what's going on there :-) Guys, just for your information, as of today, there is now an asterisk-r2 mailing list, where you may want to move this thread to (or not) :-) More and more I focus on Unicall and OpenR2 issues only, which I and other R2 users can monitor in asterisk-r2 more easily. Moy On Fri, Sep 19, 2008 at 1:10 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Yes I can see the channels with UC show channels, and it's says Idle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Friday, September 19, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Not really the unicall setup must be idem. So you can see the unicall channels ? It's moises are busy i can give you support too Regards, Luis Morales On Sat, Sep 20, 2008 at 12:15 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hi Luis, But this E1 has 30 channels, all for both directions... I must differentiate this?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Friday, September 19, 2008 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Hi dae, Your zapata.conf must be ok, now inyour unicall.conf [channels] language=es context=from-pstn usecallerid=yes hidecallerid=no immediate=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 loglevel=255 callgroup=1 pickupgroup=1 group=1 musiconhold=default relaxdtmf=yes category=NATIONAL_SUBSCRIBER ; protocolclass=mfcr2 protocolvariant=co,20,4,x,T1=1500,T2=24000,T3=15000,max-seize-wait-ack=3000 protocolend=cpe ; ; E1 IN group = 1 context = from-pstn channel = 1-15 ; E1 OUT group = 2 context = from-pstn channel = 17-31 - It's very important that you identify: - E1 lines in - E1 lines out Now from your cosole type: asterisk -r now from asterisk cli type: pbx* UC show channels The result must be an list with your unicall channels, similar to: Channel Extension Context Status Language MusicOnHold 1from-pstn Idle es default 2from-pstn Idle es default 3from-pstn Idle es default 4from-pstn Idle es default 5from-pstn Idle es default 6from-pstn Idle es default 7from-pstn Idle es default 8from-pstn Idle es default 9from-pstn Idle es default 10from-pstn Idle es default 11from-pstn Idle es default 12from-pstn Idle es default 13from-pstn Idle es default 14from-pstn Idle es default 15from-pstn Idle es default 17 6842 from-pstn Idle es default 18from-pstn Idle es default 19from-pstn Idle es default 20from-pstn Idle es default 21from-pstn Idle es default 22from-pstn Idle es default 23from-pstn Idle es default 24from-pstn Idle es default 25from-pstn Idle es default 26from-pstn Idle es default 27from-pstn Idle es default 28from-pstn Idle es default 29from-pstn Idle es default 30from-pstn
[asterisk-users] Anyone flying with BA on Monday for Astricon ?
Greetings to all ! I will be flying on Monday with BA from London to Phoenix so i was wondering if anyone else is on the same plane so there will more than inflight movies to pass the time :) If so please contact me off-list and we can arrange to meet. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to add extensions and sip registrations dynamically
Hi, I have inherited some code that appears to implement a kluge-y way of adding and removing extensions, sip devices, and sip registrations dynamically. Yep, you guessed it - it modifies the extensions.conf and sip.conf files, and then execute script to ask Asterisk to reload the dialplan and the sip module. Is there a better way to do this? :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon: Throw the dice, give a talk
Only two days until the Astricon pre-conference activities start on Tuesday, and three days until the main conference opens up in Phoenix! Many of the Digium staff are on-site already, preparing the dCAP testing facilities, getting the network ready, and sorting through the arrangements to make the conference go smoothly. For those of you who thought you couldn't make it, maybe it's time to check for those last-minute airfare deals - perhaps fate is working in your favor this week! Maybe a friend or fellow Asterisk traveller can share a room with you - ask around on IRC. With 60 (!!) talks, there is almost certainly going to be a travel problem of some type with one or two of our presenters which will lead to a speaking slot being open. With this understanding of the random nature of things, there may be some of you who will attend who have a pet project or talk you've got already written and have been really interested in presenting to an audience of Asterisk-savvy administrators, users, and business people. Here's your chance! Send along your talk idea to me via email, and I'll put it on the on deck list of emergency fill-ins for the conference. Please have a fully-fleshed out talk, including slides - while I'd like to encourage all to submit their ideas, I need a bit more detail on your preliminary proposal due to time constraints. So give it a try! Maybe fate will find fortune with you. If you're chosen as a speaker at the last minute, dinner is on me. :-) I'm looking forward to seeing everyone this year - old friends and new faces as well. Talk with you in Phoenix! JT -- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
