[asterisk-users] OT : OpenSER Summit Pavilion - 17th to 19th of March, 2008 , San Jose, US
I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of the event, see http://www.openser.org/mos/view/OpenSER-Summit-2008. All participants to register via OpenSER site before Friday, March 7th, will get free access to the OpenSER event. This OpenSER Summit edition is sponsored by a parallel OpenSER related event, the OpenSER Pavilion . The pavilion is a common exhibiting area - booth 1027, inside VoN expo, gathering, under the OpenSER name, six different companies working or using the project. You can find more about the OpenSER Pavilion and its participants here - http://www.openser.org/mos/view/OpenSER-Pavilion-2008. The dual event, OpenSER Summit Pavilion is new concept of a more complex event, aiming to create a larger diversity and to give more power to the understanding of the OpenSER project. ___ -- 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 and Cisco Unity?
Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
[asterisk-users] Friday Feb 29th Leap Year Special wih Aastra
Leap year? Election year? Will your GoToIfTime() dialplan function properly on Feb 29th? Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Aastra has some interesting phones so we asked them to come talk about them. Friday, February 29 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT * Call (724) 444-7444 or sip:[EMAIL PROTECTED] After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you have no PIN) If ( (you are registered) (PIN == callerID) ) you will not need to enter an ID; http://VoIPUsersConference.org for how to listen and join. Well known for high-quality SIP phones this presentation will be an overview of their SIP phones and explanation of their new SIP-DECT enterprise scale cordless technology. A PDF document accompanies this presentation for those who wish to follow-along Powerpoint style. http://food4wine.ning.com is the VUC Community Site (archives of all sessions are here) IRC freenode.net #voip-users-conference is the channel to ask questions if you can't call Google Group/Mailing List: http://groups.google.com/group/voip-users-conference How to set up asterisk to call in via SIP: http://voipusersconference.org/asterisktalkshoecallinsetup.htm /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] SPA3102 registration problem
hello everyone what i did to configure SPA3102 is -sip.conf- ;spa-fxs [108] type=friend host=dynamic context=sipphones secret=VerySecretPass mailbox=108 dtmfmode=rfc2833 ;dtmfmode=inband disallow=all allow=alaw ;spa-fxo-in [118] type=friend host=dynamic context=home secret=secret2 dtmfmode=rfc2833 disallow=all allow=alaw insecure=very ;spa-fxo-out [pstn-spa3k] type=peer host=XX.XX.XX.XX ; put your ip address here (i had internet address here) port=5061 secret=testingasterisk username=asterisk fromuser=asterisk dtmfmode=rfc2833 context=home insecure=very -sip.conf- don't for get to create contexts in extension.conf ---spa3102 Line 1 SIP SETTINGS: SIP Port:5060 Proxy and Registration: Proxy: YY.YY.YY.YY ; its ip address of your asterisk server Register: YES Subscriber Information: Display Name:spafxs User ID:108 Use Auth ID: NO VoIP Fallback To PSTN : Auto PSTN Fallback:YES Dial Plan: Dial Plan:([2-79]11:@gw0|xx.|*xx.|**xx.|#,:xx.:@gw0|#,:*xx:@gw0) Enable IP Dialing:NO PSTN LINE SIP SETTINGS: SIP Port:5061 Proxy and Registration: Proxy: YY.YY.YY.YY ; its ip address of your asterisk server Register: NO Subscriber Information: Display Name:CallerIDforWorld User ID:118 Use Auth ID: NO Dial Plans: Dial Plan 1:(S0:YY.YY.YY.YY); its ip address of your asterisk VoIP-To-PSTN Gateway Setup: VoIP-To-PSTN Gateway Enable:YES VoIP Caller Auth Method:None Line 1 VoIP Caller DP:None VoIP Caller Default DP:None Line 1 Fallback DP:None VoIP Users and Passwords (HTTP Authentication): VoIP User 1 Auth ID:asterisk VoIP User 1 DP:1 PSTN-To-VoIP Gateway Setup : PSTN-To-VoIP Gateway Enable:Yes PSTN Caller Auth Method:None PSTN Ring Thru Line 1:No PSTN CID For VoIP CID:Yes PSTN Caller Default DP:1 Line 1 Signal Hook Flash To PSTN:Disabled ---spa3102 I think this is enough for starting. These settings are working perfectly for my needs. (testing) On Wed, Feb 27, 2008 at 09:14:50PM +0100, Jaap Winius wrote: Quoting Tim Johnson [EMAIL PROTECTED]: I see you put a password line in your sip.conf, but I do not see a username line. Also, you might want to check the port #'s for both the Line 1 and PSTN line. I use 5060 and 5061, respectively. Hopefully this either helps, or puts you on the right track. The username is 8000, so I don't believe it's necessary to mention it. As for the ports, I'm using them in the same way you suggest. Yet it refuses to work. My first attempt involved copying my SPA3000's working configuration to the SPA3102. That didn't work. So, I reset the device and applies a configuration generated by Voxilla's wizard, which worked for me with the SPA3000. Not that this has lead to any real differences, but it's still not working. There must be something else different about the SPA3102. I did see a problem with it mentioned somewhere in which it's connection with the local Asterisk server would fail (I think temporarily) when changes to the state of its Internet connection occurred (obviously not an issue with the SPA3000). I hope this has nothing to do with my problem. Thanks anyway! Cheers, 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 -- С Уважением, Мандип Сингх Бхабха email: [EMAIL PROTECTED] ___ -- 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] New Interested services to be added for Telephoney Service Provider
Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Asterisk and Cisco Unity?
Do you mean Call Manager? Unity is just their voicemail system. Yes, you can use SIP to talk between * and CM. You can also use h.323, but it is a big hassle. Tony Mountifield wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. Cheers Tony ___ -- 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 monitor() zap channel problem
im trying to use monitor() aplication with b option, to start the recordigin just once the conversation has actuallly begun. It works fine with a sip extensión, but when i use a zap channel, it records all the channel bridging, including the ringing sounds... could you please help me with this issue? ill keep reporting 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] Coppercom and Asterisk
register = [EMAIL PROTECTED]:X:[EMAIL PROTECTED] [8159093010] fromdomain=proxy.essex1.com host=proxy.essex1.com port=5060 insecure=very username=8159093010 secret=X type=peer qualify=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=ulaw outboundproxy=proxy.essex1.com [Feb 28 07:44:52] NOTICE[9409]: chan_sip.c:7364 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]@proxy.essex1.com' timed out, trying again (Attempt #1) REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 63.164.210.14:5060: REGISTER sip:proxy.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK24661abe;rport From: sip:[EMAIL PROTECTED];tag=as16c1714c To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Aiur*CLI --- SIP read from 63.164.210.14:5060 --- SIP/2.0 423 Interval Too Brief To: sip:[EMAIL PROTECTED];tag=ddcdjfgdeigdhifj-bibgaceacb From: sip:[EMAIL PROTECTED];tag=as16c1714c Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK24661abe Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Expires: 120 Min-Expires: 900 Content-Length: 0 - --- (9 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from 63.164.210.14 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Feb 28 07:45:12] NOTICE[9409]: chan_sip.c:7364 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]@proxy.essex1.com' timed out, trying again (Attempt #2) REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 63.164.210.14:5060: REGISTER sip:proxy.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK7db9ed82;rport From: sip:[EMAIL PROTECTED];tag=as4a12e1ea To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Aiur*CLI --- SIP read from 63.164.210.14:5060 --- SIP/2.0 423 Interval Too Brief To: sip:[EMAIL PROTECTED];tag=ejhgidfdeiidhifj-bacgaceacb From: sip:[EMAIL PROTECTED];tag=as4a12e1ea Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK7db9ed82 Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER Expires: 120 Min-Expires: 900 Content-Length: 0 - --- (9 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from 63.164.210.14 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Wednesday, February 20, 2008 4:52 PM Subject: [asterisk-users] Coppercom and Asterisk My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - X No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use outbound Proxy -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.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___ -- 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] Friday Feb 29th Leap Year Special wih Aastra
On Thursday 28 February 2008 05:13:06 randulo wrote: Will your GoToIfTime() dialplan function properly on Feb 29th? It will work fine. In fact, you can put February 30th or February 31st into your GotoIfTime arguments, and it will accept the values just fine (it just won't ever evaluate true). -- Tilghman ___ -- 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] New Interested services to be added for Telephoney Service Provider
Do you have an English translation of this post? On Thu, Feb 28, 2008 at 6:48 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Simultaneous Inbound and Outbound calls on analog lines...
In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. HTH, James Texter On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- 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] Simultaneous Inbound and Outbound calls on analog lines...
On Thu, Feb 28, 2008 at 08:47:37AM -0600, James Texter III wrote: On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. And, to expand a bit, this comes from the fact that on analog (and I guess RBT-1, though I hadn't realized it happened there), there is not a 3-way handshake to open the channel; each end can open it unilaterally. This leads to a race condition, and thus, 'glare'. The *reliable* way to fix this is to go to PRI, if you can a) get it, b) afford it, c) terminate it, and d) profit!! Oh, sorry; that's Slashdot. Never mind. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Simultaneous Inbound and Outbound calls on analog lines...
Thank you all for the suggestions. I'm looking into getting groundstart lines for that installation as suggested earlier. Also, I'll try setting the outbound call routes in reverse from the inbound hunt group. I appreciate your help! Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: James Texter III [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 28, 2008 8:47:37 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines... In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. HTH, James Texter On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- 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] Friday Feb 29th Leap Year Special wih Aastra
On Thu, Feb 28, 2008 at 3:16 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Thursday 28 February 2008 05:13:06 randulo wrote: Will your GoToIfTime() dialplan function properly on Feb 29th? It will work fine. In fact, you can put February 30th or February 31st into your GotoIfTime arguments, and it will accept the values just fine (it just won't ever evaluate true). Doesn't that depend on what planet you are currently on? Or like, if you were in ancient Rome or something, with a Cesarian calendar? ___ -- 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] Simultaneous Inbound and Outbound calls on analog lines...
:-) HAHA.. Unfortunately, PRI service is not available at this location... Thank you for the help! Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Jay R. Ashworth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 28, 2008 8:59:29 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines... On Thu, Feb 28, 2008 at 08:47:37AM -0600, James Texter III wrote: On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. And, to expand a bit, this comes from the fact that on analog (and I guess RBT-1, though I hadn't realized it happened there), there is not a 3-way handshake to open the channel; each end can open it unilaterally. This leads to a race condition, and thus, 'glare'. The *reliable* way to fix this is to go to PRI, if you can a) get it, b) afford it, c) terminate it, and d) profit!! Oh, sorry; that's Slashdot. Never mind. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Digium certified asterisk professional linkedin group
Dear all, I've created a digium certified asterisk professional - dCAP linkedin group for anyone, dCAP, interested: http://www.linkedin.com/e/gis/60298/39AE1350DBF3 Best regards, Marco Mouta dCAP November 2006 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. 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] Unicall mfcr2 testcall issues in mexico outgoing:ok | incoming: fail.
This may fix your issue: mx,10,4,0 By default Mexico variant has the option get ANI after DNIS. Which it means just after getting the DNIS digits we will request the calling party category and DNIS. The Nortel PBX seems to not like calling party category requests and they want to go straight to group II signal instead of group C. Adding a 0 as options will disable the get ANI after DNIS option and go straight to Group II signals. Give that a try and let us know, tho, I still wonder why the Nortel does not accept the Calling Party Category Request and Switch to Group C signal. Is the Nortel PBX properly configured for México variant??? Moisés Silva On Wed, Feb 27, 2008 at 8:56 PM, Andres Tello Abrego [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Thanks Carlos... Using mx,10,4 didn't work. Chan 31, class 'mfcr2', variant 'mx,10,4', end 0, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 31: Call control(9) MFC/R2 Chan 31: Unblock MFC/R2 Chan 31: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 31: far_unblocking_expired MFC/R2 Chan 31: local_unblocking_expired Chan 31: -- Far end unblocked! :-) Chan 31: -- Far end unblocked! :-) Chan 31: -- Local end unblocked! :-) Chan 31: -- Local end unblocked! :-) MFC/R2 Chan 31: - 0001 [1/IDLE/Idle /Idle ] MFC/R2 Chan 31: Detected MFC/R2 Chan 31: Creating a new call with CRN 32769 MFC/R2 Chan 31: 1101 - [2/DETECTED/Seize ack /Seize ack] Chan 31: -- Detected on channel 0, CRN 32769 Chan 31: -- Detected on channel 0, CRN 32769 Main thread MFC/R2 Chan 31: - 8 on [2/DETECTED/Seize ack /Seize ack] MFC/R2 Chan 31: 1 on - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 8 off [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 off - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 6 on [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 on - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 6 off [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 off - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 1 on [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 on - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 1 off [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 off - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 0 on [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 6 on - [2/DETECTED/Group C /Category req ] MFC/R2 Chan 31: - 0 off [2/DETECTED/Group C /Category req ] MFC/R2 Chan 31: 6 off - [2/DETECTED/Group C /Category req ] Main thread Main thread Main thread MFC/R2 Chan 31: R2 prot. err. [2/DETECTED/Group C /Category req ] cause 32771 - T3 timed out MFC/R2 Chan 31: 1001 - [1/IDLE/Idle /Idle ] Chan 31: -- Protocol failure on channel 0, cause (32771) T3 timed out Chan 31: -- Protocol failure on channel 0, cause (32771) T3 timed out MFC/R2 Chan 31: - 1001 [1/IDLE/Idle /Idle ] MFC/R2 Chan 31: 1001 - [1/IDLE/Idle /Idle ] Carlos Chavez wrote: I do not know if this will make a difference but the protocol-variant for Mexico should be: protocol-variant mx,10,4 You only get 10 digits from the phone company. On Wed, 2008-02-27 at 18:03 -0800, Andres Tello Abrego wrote: protocol-variant mx,20,4 ___ -- 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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) Comment: Using GnuPG with SUSE - http://enigmail.mozdev.org iD8DBQFHxiLHEXCJrml2yYoRAt5+AKCOXNfIUZYDDpGb0jSBO2Ulz4q+fgCbBBum Ux+Q+w33ZGgtApwNOZWOLGA= =IksU -END PGP 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] C Code to connect to Asterisk Manager Interface
Hi, I believe your problem of authorization is relative to astersik's
manager.conf configuration and you need to add and user and password in the
manager.conf to have remote access. I have used some examples of voip-info.org,
look at this link in the second half part, it explain how to configure the
manager.conf file.
http://www.voip-info.org/wiki/view/Asterisk%20Zaptel%20Nagios%20plugin
# You need a manager entry in /etc/asterisk/manager.conf
# [nagios]
# secret=somesecret
# deny=0.0.0.0/0.0.0.0
# permit=127.0.0.0/255.0.0.0
# permit=111.222.333.444/255.255.255.111 -- the network nagios connects from
# read = system,call,log,verbose,command,agent,user
# write = system,call,log,verbose,command,agent,user
and in this other link, you can find the Manager API and another examples:
http://www.voip-info.org/wiki/view/Asterisk+manager+API
regards,
Claro Taroco
- Mensaje original
De: Michael Henderson [EMAIL PROTECTED]
Para: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Enviado: jueves 28 de febrero de 2008, 5:14:36
Asunto: [asterisk-users] C Code to connect to Asterisk Manager Interface
Hi,
I have written a C code which would let me connect to the Asterisk Manager
Interface. The code compiles successfully but on running the code I get
unauthorized login shown in the Asterisk command line console.
Here is my C code:
#includestdio.h
#includenetdb.h
#includeunistd.h
#includestring.h
#includearpa/inet.h
#includesys/types.h
#includesys/socket.h
#includenetinet/in.h
#define MAX_MSG_SIZE 512
#define SERVER_ADDRESS 192.168.0.150
#define CLIENT_ADDRESS 192.168.0.150
#define SERVER_PORT 5038
#defineCLIENT_PORT 5100
int main()
{
int sd;
struct sockaddr_in serveraddr, clientaddr;
char msg[MAX_MSG_SIZE];
bzero((char *) serveraddr, sizeof(serveraddr));
serveraddr.sin_family = AF_INET;
serveraddr.sin_addr.s_addr = inet_addr(SERVER_ADDRESS);
serveraddr.sin_port = htons(SERVER_PORT);
bzero((char *) clientaddr, sizeof(clientaddr));
clientaddr.sin_family = AF_INET;
clientaddr.sin_addr.s_addr = INADDR_ANY;
clientaddr.sin_port = htons(CLIENT_PORT);
sd = socket(AF_INET, SOCK_STREAM, 0);
printf(\nCreated socket ...);
bind(sd,(struct sockaddr *) clientaddr, sizeof(clientaddr));
printf(\nBinding successful ...);
connect(sd,(struct sockaddr *) serveraddr, sizeof(serveraddr));
printf(\nConnected ...);
*msg=(char)Action: Login\r\nUsername: admin\r\nSecret: admin\r\nActionID:
1\r\n\r\n;
send(sd,msg,strlen(msg)+1,0);
close(sd);
return(1);
}
Please correct me where I am going wrong. In manager.conf the username and
secret has been defined.
Thank you.
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Re: [asterisk-users] New Interested services to be added for Telephoney Service Provider
I'm pretty sure he's asking what sort of advantages there are in using VoIP (and probably Asterisk) over traditional wireline services. Advantages being things like flexibility and portability (with cost and barriers-to-entry being somewhat debatable). But he's more interested perhaps in the technology side specifically? Don't know. Bilal? Can you be more specific? N. C F wrote: Do you have an English translation of this post? On Thu, Feb 28, 2008 at 6:48 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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
[asterisk-users] quickfix for building zaptel with 2.6.24?
Hi, I am trying to build zaptel 1.4.8 with kernel 2.6.24 on debian/sid: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? 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] Asterisk monitor() zap channel problem
Raul Alarcon wrote: im trying to use monitor() aplication with b option, to start the recordigin just once the conversation has actuallly begun. It works fine with a sip extensión, but when i use a zap channel, it records all the channel bridging, including the ringing sounds... could you please help me with this issue? ill keep reporting thanks. I think it is because analog lines to not provide call progress like sip does. Someone more knowledgeable can correct me here if I'm wrong, but that is my first guess. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- 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] Unicall mfcr2 testcall issues in mexico outgoing:ok | incoming: fail.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Same effect... I belive that is a nortel issue. But I have no idea of how to debug it to fix it... any advice is helped.. Also the provider, asked me for the tone table because he can set the tone table as he wishes... TIA. Testcalll output ./testcall Chan 10, class 'mfcr2', variant 'mx,10,4,0', end 0, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 10: Call control(9) MFC/R2 Chan 10: Unblock MFC/R2 Chan 10: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 10: far_unblocking_expired MFC/R2 Chan 10: local_unblocking_expired Chan 10: -- Far end unblocked! :-) Chan 10: -- Far end unblocked! :-) Chan 10: -- Local end unblocked! :-) Chan 10: -- Local end unblocked! :-) MFC/R2 Chan 10: - 0001 [1/IDLE/Idle /Idle ] MFC/R2 Chan 10: Detected MFC/R2 Chan 10: Creating a new call with CRN 32769 MFC/R2 Chan 10: 1101 - [2/DETECTED/Seize ack /Seize ack] Chan 10: -- Detected on channel 0, CRN 32769 Chan 10: -- Detected on channel 0, CRN 32769 MFC/R2 Chan 10: - 8 on [2/DETECTED/Seize ack /Seize ack] MFC/R2 Chan 10: 6 on - [2/DETECTED/Group C /Category req ] MFC/R2 Chan 10: - 8 off [2/DETECTED/Group C /Category req ] MFC/R2 Chan 10: 6 off - [2/DETECTED/Group C /Category req ] Main thread MFC/R2 Chan 10: R2 prot. err. [2/DETECTED/Group C /Category req ] cause 32771 - T3 timed out MFC/R2 Chan 10: 1001 - [1/IDLE/Idle /Idle ] Chan 10: -- Protocol failure on channel 0, cause (32771) T3 timed out Chan 10: -- Protocol failure on channel 0, cause (32771) T3 timed out Main thread MFC/R2 Chan 10: - 1001 [1/IDLE/Idle /Idle ] MFC/R2 Chan 10: 1001 - [1/IDLE/Idle /Idle ] Main thread Moises Silva wrote: This may fix your issue: mx,10,4,0 By default Mexico variant has the option get ANI after DNIS. Which it means just after getting the DNIS digits we will request the calling party category and DNIS. The Nortel PBX seems to not like calling party category requests and they want to go straight to group II signal instead of group C. Adding a 0 as options will disable the get ANI after DNIS option and go straight to Group II signals. Give that a try and let us know, tho, I still wonder why the Nortel does not accept the Calling Party Category Request and Switch to Group C signal. Is the Nortel PBX properly configured for México variant??? Moisés Silva On Wed, Feb 27, 2008 at 8:56 PM, Andres Tello Abrego [EMAIL PROTECTED] wrote: Thanks Carlos... Using mx,10,4 didn't work. Chan 31, class 'mfcr2', variant 'mx,10,4', end 0, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 31: Call control(9) MFC/R2 Chan 31: Unblock MFC/R2 Chan 31: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 31: far_unblocking_expired MFC/R2 Chan 31: local_unblocking_expired Chan 31: -- Far end unblocked! :-) Chan 31: -- Far end unblocked! :-) Chan 31: -- Local end unblocked! :-) Chan 31: -- Local end unblocked! :-) MFC/R2 Chan 31: - 0001 [1/IDLE/Idle /Idle ] MFC/R2 Chan 31: Detected MFC/R2 Chan 31: Creating a new call with CRN 32769 MFC/R2 Chan 31: 1101 - [2/DETECTED/Seize ack /Seize ack] Chan 31: -- Detected on channel 0, CRN 32769 Chan 31: -- Detected on channel 0, CRN 32769 Main thread MFC/R2 Chan 31: - 8 on [2/DETECTED/Seize ack /Seize ack] MFC/R2 Chan 31: 1 on - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 8 off [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 off - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 6 on [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 on - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 6 off [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 off - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 1 on [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 on - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 1 off [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 1 off - [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: - 0 on [2/DETECTED/Group A /DNIS request ] MFC/R2 Chan 31: 6 on - [2/DETECTED/Group C /Category req ] MFC/R2 Chan 31: - 0 off [2/DETECTED/Group C /Category req ] MFC/R2 Chan 31: 6 off - [2/DETECTED/Group C /Category req ] Main thread Main thread Main thread MFC/R2 Chan 31: R2 prot. err. [2/DETECTED/Group C /Category req ] cause 32771 - T3 timed out MFC/R2 Chan 31: 1001 - [1/IDLE/Idle /Idle ] Chan 31: --
Re: [asterisk-users] Asterisk and Cisco Unity?
Tony wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. As Peder mentioned, Unity is only a VM platform. I actually started using Asterisk to replace a Cisco Conferencing package that never worked right. We have had it running internally for three+ years now, and have been very happy with the results. I am currently using chan_ooh323, but SIP is possible if you have CCM 4.2 or higher. You'll also want to run a later release of Asterisk 1.4 which has a work-around for an odd CCM hold implementation. 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] quickfix for building zaptel with 2.6.24?
Louis-David Mitterrand wrote: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? Yes, use Zaptel 1.4.9.1 or wait for the release of 1.4.10 later today or first thing tomorrow. If you decide to use 1.4.9.1, please note that if you are using analog cards with FXO modules, there is a known bug in DTMF generation that will affect your ability to dial out on those ports. That has been fixed in Subversion (see issue 11855 on bugs.digium.com) and will be in the next release. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] OT But I Would Rather See People Running Asterisk on a Real Server than an Emachine
Subject: I Would Rather See People Running Asterisk on a Real Server than an Emachine It's funny you say that - I've got a 667MHz eMachine with 256mb of RAM running Trixbox + hylafax/iaxmodem, routing our Internet traffic w/iptables, proxying the kids' net traffic w/squid, samba... dhcpd... cvsd... bind... It's chugging along quite nicely. That said, this is our home setup. I have a couple of small side businesses (technical consulting astronomy accessories) which it's handling the phones for, but our main business is being run off a Dell PE 6650. St- ___ -- 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] Problems with setting up Zaptel
Hi all, I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just trying to get it working. But no luck as of yet. In /etc/zaptel.conf I've set the following options: fxsks=2 fxoks=1 loadzone=se defaultzone=se And in /etc/asterisk/zapata.conf I've not sure what to set exactly. For example, under [trunkgroups] what to specify there? Under [channels I set something like the following: usecallerid=yes cidsignalling=dtmf cidstart=polarity hanguponpolarityswitch=yes signalling=fxs_ks context=default Any help would a be appreciated, many thanks! Christian ___ -- 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] New Interested services to be added for Telephoney Service Provider
Hi; Yes what u said is correct, I am interested in using VoIP and Asterisk (also) over wireline services (telephon line). Actually the service provider company asking for such things to be added with the telephone lines that they give it for their customer. Actually they build 9 PSTN in the state, and they are now asking for new features (advantages) that can be offered as new idea in VoIP and Asterisk. Am still not clear? Any advise and sharing ideas can be offered? I already told about Virual PBX and Virual Contact Center and having VoIP numbers to be offered with the telephone lines, but I do not know if Asterisk have something good that can offer in Virual PBX and Virual Contact Center or portability, also any other new ideas to be integrated with the PSTN's will be welcomed. Regards Bilal I'm pretty sure he's asking what sort of advantages there are in using VoIP (and probably Asterisk) over traditional wireline services. Advantages being things like flexibility and portability (with cost and barriers-to-entry being somewhat debatable). But he's more interested perhaps in the technology side specifically? Don't know. Bilal? Can you be more specific? N. C F wrote: Do you have an English translation of this post? On Thu, Feb 28, 2008 at 6:48 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Asterisk and Cisco Unity?
Thanks for the info, Dan Peder. It helps me to know the right questions to ask the customer! Cheers Tony In article [EMAIL PROTECTED], Dan Austin [EMAIL PROTECTED] wrote: Tony wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. As Peder mentioned, Unity is only a VM platform. I actually started using Asterisk to replace a Cisco Conferencing package that never worked right. We have had it running internally for three+ years now, and have been very happy with the results. I am currently using chan_ooh323, but SIP is possible if you have CCM 4.2 or higher. You'll also want to run a later release of Asterisk 1.4 which has a work-around for an odd CCM hold implementation. 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 -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
arkda wrote: Nothing in the console aside from what I've posted. When a DTMF tone is played the server freezes instantly, hard reboot required. Just to close out this thread, it appears that this issue was related to http://bugs.digium.com/view.php?id=12053 Adding a loadzone and defaultzone to the /etc/zaptel.conf file resolved the server freeze/ crash. ___ -- 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] Asterisk and Cisco Unity?
Hello , about this implementacion , i have a issue with ASterisk 1.4.2 and Cisco Unity , the VM doesn't work fine the calls are good but when enter the VM ( cisco Unity ) it didn't work . Somebody has one implementacion ? To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Thu, 28 Feb 2008 20:35:09 + Subject: Re: [asterisk-users] Asterisk and Cisco Unity? Thanks for the info, Dan Peder. It helps me to know the right questions to ask the customer! Cheers Tony In article [EMAIL PROTECTED], Dan Austin [EMAIL PROTECTED] wrote: Tony wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure.As Peder mentioned, Unity is only a VM platform. I actually started using Asterisk to replace a Cisco Conferencing package that never worked right. We have had it running internally for three+ years now, and have been very happy with the results.I am currently using chan_ooh323, but SIP is possible if you have CCM 4.2 or higher. You'll also want to run a later release of Asterisk 1.4 which has a work-around for an odd CCM hold implementation.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 -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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 _ Tecnología, moda, motor, viajes,…suscríbete a nuestros boletines para estar siempre a la última http://newsletters.msn.com/hm/maintenanceeses.asp?L=ESC=ESP=WCMaintenanceBrand=WLRU=http%3a%2f%2fmail.live.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] New Interested services to be added forTelephoney Service Provider
I think Bilal's service provider is asking What is the next Killer Ap for VoIP? --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Thursday, February 28, 2008 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Interested services to be added forTelephoney Service Provider Hi; Yes what u said is correct, I am interested in using VoIP and Asterisk (also) over wireline services (telephon line). Actually the service provider company asking for such things to be added with the telephone lines that they give it for their customer. Actually they build 9 PSTN in the state, and they are now asking for new features (advantages) that can be offered as new idea in VoIP and Asterisk. Am still not clear? Any advise and sharing ideas can be offered? I already told about Virual PBX and Virual Contact Center and having VoIP numbers to be offered with the telephone lines, but I do not know if Asterisk have something good that can offer in Virual PBX and Virual Contact Center or portability, also any other new ideas to be integrated with the PSTN's will be welcomed. Regards Bilal I'm pretty sure he's asking what sort of advantages there are in using VoIP (and probably Asterisk) over traditional wireline services. Advantages being things like flexibility and portability (with cost and barriers-to-entry being somewhat debatable). But he's more interested perhaps in the technology side specifically? Don't know. Bilal? Can you be more specific? N. C F wrote: Do you have an English translation of this post? On Thu, Feb 28, 2008 at 6:48 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Simultaneous Inbound and Outbound calls on analog lines...
Tim Nelson wrote: Thank you all for the suggestions. I'm looking into getting groundstart lines for that installation as suggested earlier. Make sure your interface supports GS The Sangoma and TDM cards do I assume you are using one of these as you mention Zaptel. John Novack Also, I'll try setting the outbound call routes in reverse from the inbound hunt group. I appreciate your help! Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: James Texter III [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 28, 2008 8:47:37 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines... In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. HTH, James Texter On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- 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 -- Dog is my co-pilot ___ -- 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] Simultaneous Inbound and Outbound calls on analog lines...
Yes... this installation has a Sangoma A400D card fully populated. Thanks again. Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 28, 2008 3:51:47 PM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines... Tim Nelson wrote: Thank you all for the suggestions. I'm looking into getting groundstart lines for that installation as suggested earlier. Make sure your interface supports GS The Sangoma and TDM cards do I assume you are using one of these as you mention Zaptel. John Novack Also, I'll try setting the outbound call routes in reverse from the inbound hunt group. I appreciate your help! Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: James Texter III [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 28, 2008 8:47:37 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines... In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. HTH, James Texter On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- 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 -- Dog is my co-pilot ___ -- 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] GLOBAL function - introduced at what version?
I understand the use of the g option in a call of Set() is deprecated as of version 1.4. Was the GLOBAL function used to replace it introduced in version 1.4 or were there some late 1.2 versions that also supported it? -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ -- 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 Voicemail for iPhone
Heres a little teaser for those of you with iPhones Asterisk Voicemail for iPhone allows you to check your voicemail messages on your house or business line from your iPhone. You can think of it as Visual Voicemail, but for your Asterisk PBX numbers instead of your ATT cell number. The technology behind it is Asterisk (The Open-Source PBX), with iUI, Joe Hewitt's UI interface for iPhone. This software can be installed on any Asterisk server (though you will want to use one that is available via the Internet) and will allow you to check messages in multiple folders, listen to messages, delete messages, move messages, and change voicemail settings - all from your iPhone. Contact me with any questions or comments. This software is unreleased. Most of the features are fully functional, but I need to clean up certain portions of the code before releasing it in order to avoid public ridicule. This software will be released under the GPL or some other free license. http://chriscarey.com/projects/asterisk/iphone/ ___ -- 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] GLOBAL function - introduced at what version?
On Thursday 28 February 2008 16:04:49 Phil Reynolds wrote: I understand the use of the g option in a call of Set() is deprecated as of version 1.4. Was the GLOBAL function used to replace it introduced in version 1.4 or were there some late 1.2 versions that also supported it? It was introduced in 1.4. In both 1.2 and 1.4, we have a policy of no new features, so the change was not eligible for 1.2. -- Tilghman ___ -- 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] TDM400P dialout problem
Anthony Messina wrote: Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: This should be fixed in Zaptel 1.4.9.2. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] Asterisk Voicemail for iPhone
This looks great, Cant wait to try it on my iphone -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Carey Sent: Friday, 29 February 2008 9:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Voicemail for iPhone Heres a little teaser for those of you with iPhones Asterisk Voicemail for iPhone allows you to check your voicemail messages on your house or business line from your iPhone. You can think of it as Visual Voicemail, but for your Asterisk PBX numbers instead of your ATT cell number. The technology behind it is Asterisk (The Open-Source PBX), with iUI, Joe Hewitt's UI interface for iPhone. This software can be installed on any Asterisk server (though you will want to use one that is available via the Internet) and will allow you to check messages in multiple folders, listen to messages, delete messages, move messages, and change voicemail settings - all from your iPhone. Contact me with any questions or comments. This software is unreleased. Most of the features are fully functional, but I need to clean up certain portions of the code before releasing it in order to avoid public ridicule. This software will be released under the GPL or some other free license. http://chriscarey.com/projects/asterisk/iphone/ ___ -- 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 -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, 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
[asterisk-users] Problems with removing zaptel
Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel modules some of them cannot be removed. The other module is in use. Also if I reboot my system they're all loaded again. Any thoughts? Many thanks, Christian ___ -- 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] TDM400P dialout problem
Is this only on the _64 zaptel or will affect ALL zpatel 1.4.9 ? -Original Message- From: Russell Bryant [EMAIL PROTECTED] Sent: Feb 28, 2008 6:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM400P dialout problem Anthony Messina wrote: Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: This should be fixed in Zaptel 1.4.9.2. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] TDM400P dialout problem
On Thursday 28 February 2008 05:41:55 pm Al Baker wrote: Is this only on the _64 zaptel or will affect ALL zpatel 1.4.9 ? -Original Message- From: Russell Bryant [EMAIL PROTECTED] Sent: Feb 28, 2008 6:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM400P dialout problem Anthony Messina wrote: Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: This should be fixed in Zaptel 1.4.9.2. thanks russell. in reply to al: with 1.4.7.1, i had no problems with either x86_64 or i386. with 1.4.8, i386 worked, but x86_64 did not. with 1.4.9 and 1.4.9.1, neither worked. i use the rpms from atrpms.net for fedora 7 i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. unfortunately, due to my work schedule, i did not have time to debug the differences between the platforms. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Problems with removing zaptel
/etc/modprobe/blacklistor similar PaulH On Fri, 2008-02-29 at 00:30 +0100, Christian wrote: Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel modules some of them cannot be removed. The other module is in use. Also if I reboot my system they're all loaded again. Any thoughts? Many thanks, Christian ___ -- 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] TDM400P dialout problem
Anthony Messina wrote: i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. kpfleming has done some work today on this issue which needs a little more in house testing. In the interim, with the current version of the wctdm driver, you can pass the battdebounce as a module parameter. The units in the wctdm driver are 16ms. So for example, you could try passing battdebounce=12 to set up a 200ms debounce and see if that does the trick for you depending on your location. ___ -- 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] Friday Feb 29th Leap Year Special wih Aastra
If anyone has managed to compile and run Asterisk on a server from this particular era, I'd /love/ to know about it. :) What's the performance like? For that matter, what phones were available at the time? randulo wrote: On Thursday 28 February 2008 05:13:06 randulo wrote: Will your GoToIfTime() dialplan function properly on Feb 29th? It will work fine. In fact, you can put February 30th or February 31st into your GotoIfTime arguments, and it will accept the values just fine (it just won't ever evaluate true). Doesn't that depend on what planet you are currently on? Or like, if you were in ancient Rome or something, with a Cesarian calendar? ___ -- 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] load balancing
Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You 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
Re: [asterisk-users] New Interested services to be added forTelephoney Service Provider
Dean Collins will sell you ideas. On Thu, Feb 28, 2008 at 4:32 PM, Don Kelly [EMAIL PROTECTED] wrote: I think Bilal's service provider is asking What is the next Killer Ap for VoIP? --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Thursday, February 28, 2008 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Interested services to be added forTelephoney Service Provider Hi; Yes what u said is correct, I am interested in using VoIP and Asterisk (also) over wireline services (telephon line). Actually the service provider company asking for such things to be added with the telephone lines that they give it for their customer. Actually they build 9 PSTN in the state, and they are now asking for new features (advantages) that can be offered as new idea in VoIP and Asterisk. Am still not clear? Any advise and sharing ideas can be offered? I already told about Virual PBX and Virual Contact Center and having VoIP numbers to be offered with the telephone lines, but I do not know if Asterisk have something good that can offer in Virual PBX and Virual Contact Center or portability, also any other new ideas to be integrated with the PSTN's will be welcomed. Regards Bilal I'm pretty sure he's asking what sort of advantages there are in using VoIP (and probably Asterisk) over traditional wireline services. Advantages being things like flexibility and portability (with cost and barriers-to-entry being somewhat debatable). But he's more interested perhaps in the technology side specifically? Don't know. Bilal? Can you be more specific? N. C F wrote: Do you have an English translation of this post? On Thu, Feb 28, 2008 at 6:48 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] load balancing
On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL PROTECTED] wrote: Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I can think of would be to forward an incoming call for a user to all 4 of your Asterisk servers that way the call will get through but if they are not behind NAT they'll get 4 incoming calls. Bascially it's messy using the set up you've got. What you really need is a SIP Proxy (assuming you're using SIP, if not it's even trickier). The SIP Proxy could load balance requests across your Asterisk servers. For calls destined for your users you can use the outboundproxy field in the sippeers table, by setting it to the IP address of your SIP Proxy server you can get Asterisk to forward all requests for a SIP account through the proxy (there is also an outboundproxyport setting but avoid it as it's been broken forever). There are a few gotchas with a SIP Proxy the main one being transfers. But if you can get away with not allowing transfers then you are best to do so as the CDR's Asterisk produces are wrong anyway. Regards, Greyman. ___ -- 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] 1EZphone is only two way browser softphone - SIP Softphones and Citrix ?
Yes, try http://1ezphone.com its a browser softphone. - Original Message - From: Zoa To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Softphones and Citrix ? Date: Fri, 01 Feb 2008 23:09:56 +0200 I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something that seems to be doable at this time. I have to assume that the issues affecting the virtualization of cisco softphones with Citrix will come into play with SIP softphones as well. Seems that the two biggest issues revolve around wrapping the UDP stream up with the ICA protocol, and possibly issues with the various mics and speakers and having to interface with them I think. However, I am also a firm believer that anything is possible, practical well not usually, and it may just be the time has not come yet for this. There is a good article about this over at: http://www.brianmadden.com/content/article/How-should-Citrix-integrate-VoIP-with-Presentation-Server Any thoughts, comments or insight into this and your experiences around any of this is appreciated. ___ -- 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 -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.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
[asterisk-users] I would like to hire someone to automate my asterisk for hosted PBX service
I would like to hire someone to automate my asterisk for hosted PBX
service for fetures like user signup, adding money and call bridging
Please contact me offline at [EMAIL PROTECTED]
- Original Message -
From: Philipp Kempgen
To: Asterisk Users
Subject: Re: [asterisk-users] Running AGI script if condition met?
Date: Thu, 06 Dec 2007 05:11:24 +0100
Vincent wrote:
exten = 777,n,ExecIf($[${LEN(${CALLERIDNUM})} =
10],AGI(/root/dummy.php),${CALLERIDNUM})
The line break is not a good idea.
It doesn't look like ExecIf() is the right way to have Asterisk run
an
AGI script conditionnally. What would be the right way to do this?
Wrong syntax.
ExecIf(||)
So:
ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI,/root/dummy.php)
Not sure about more than one argument. Maybe
ExecIf($[${LEN(${CALLERIDNUM})} =
10],AGI,/root/dummy.php,${CALLERIDNUM})
or
ExecIf($[${LEN(${CALLERIDNUM})} =
10],AGI,/root/dummy.php|${CALLERIDNUM})
Asterisk's syntax is strange sometimes ...
Regards,
Philipp Kempgen
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Re: [asterisk-users] load balancing
Hi Greyman, Should it look like this now? Can i use 2 SIP Proxies just to make sure i have a backup? will that cause any problem again with regards to calling extension to extension? Extensions will register on the asterisk still? How about outbound calls to other SIP provider or a telcobridge, SIP proxy will handle that also? Basically asterisk will ask SIP proxy of everything? Will RTP stream still go thru asterisk? Also, i plan on setting these up as a Virtual PBX for multiple offices, which means company A can only use Trunks for A, B is for Trunk B etc etc. Does outbound to trunks have any issues? or problem is just basically calling extension to extension? [other voip provider][telcobridge] -- [pstn] || [ SIP Proxy ] | | | | [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | | mysql cluster| Thanks Regards, Ron Grey Man wrote: On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL PROTECTED] wrote: Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I can think of would be to forward an incoming call for a user to all 4 of your Asterisk servers that way the call will get through but if they are not behind NAT they'll get 4 incoming calls. Bascially it's messy using the set up you've got. What you really need is a SIP Proxy (assuming you're using SIP, if not it's even trickier). The SIP Proxy could load balance requests across your Asterisk servers. For calls destined for your users you can use the outboundproxy field in the sippeers table, by setting it to the IP address of your SIP Proxy server you can get Asterisk to forward all requests for a SIP account through the proxy (there is also an outboundproxyport setting but avoid it as it's been broken forever). There are a few gotchas with a SIP Proxy the main one being transfers. But if you can get away with not allowing transfers then you are best to do so as the CDR's Asterisk produces are wrong anyway. Regards, Greyman. ___ -- 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] load balancing
If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I See the discussion a few days ago. The Asterisk server saves the value of SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL. Regards, __Yehavi: ___ -- 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] load balancing
On Fri, Feb 29, 2008 at 4:03 AM, Ron [EMAIL PROTECTED] wrote: Hi Greyman, Should it look like this now? Can i use 2 SIP Proxies just to make sure i have a backup? will that cause any problem again with regards to calling extension to extension? Extensions will register on the asterisk still? How about outbound calls to other SIP provider or a telcobridge, SIP proxy will handle that also? Basically asterisk will ask SIP proxy of everything? Will RTP stream still go thru asterisk? Also, i plan on setting these up as a Virtual PBX for multiple offices, which means company A can only use Trunks for A, B is for Trunk B etc etc. Does outbound to trunks have any issues? or problem is just basically calling extension to extension? [other voip provider][telcobridge] -- [pstn] || [ SIP Proxy ] | | | | [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | | mysql cluster| Thanks Regards, Ron Hi Ron, Yep it starts to get tricky :). There will be slight difference depending exactly on what you need to accomplish. I work for a VoIP Proivder that provides services to users in internet land so our set up is designed for that. If you've got VPNs or are on a LAN things will be different. Two SIP Proxy's are definitely a good idea, you can load balance your users across them using DNS SRV records, DNS Round Robin, IP Load Balancer (although then you prob should have two load balancers). If you're just starting your build network build or only have 1000 users the extra SIP Proxy should go to the bottom of the list. SIP Proxy's such as OpenSER are pretty stable and don't do anywhere near as much work as the media server. It's the fault tolerance on your Asterisk servers that is the most critical thing. They do a lot more work and in my experience with them (4+ years) they are a lot more likely to crash than your SIP Proxy. With two SIP Proxy's you have an additional problem in that now you need to set the outboundproxy field in the Asterisk realtime database to the value of the proxy the user agent came through. Asterisk can't do that for you (as far as I know) so you could possibly use the SIP Proxy to do registrations or use a custom SIP Registrar. Both are a good idea as they take registration load away from Asterisk and this can be VERY significant as your user base grows. We use a custom SIP Registrar. For outbound trunking we go directly from Asterisk to the terminating gateway no SIP Proxy involved. For inbound trunking we do go through the SIP Proxy for the same reason you get users to. Incoming calls are going to be more reliable if they are not tied to a single Asterisk server (I guess you could use SRV records for your Asterisk servers for inbound trunking as well but then you're kind of duplicating the role of the SIP proxy). The RTP stream will always be between the users and Asterisk the SIP Proxy is never invovled. If you send an RTP packet to a SIP Proxy and it will just shake its head in an irritated manner and ignore you. Regards, Greyman. ___ -- 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] load balancing
On Fri, 29 Feb 2008 6:21 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: See the discussion a few days ago. The Asterisk server saves the value of SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL. Regards, __Yehavi: Ahh that's handy. That would allow a half way solution between multiple Asterisk servers and a SIP Proxy by utilising an AGI script or database lookup in each Asterisk server's dialplan. When the incoming calls arrive you'll be able to know which Asterisk server to forward them to. You still have the problems about failing over the Asterisk servers and putting two Asterisk servers in the media path is always best avoided if possible although probably not a huge deal. Actually from memory there is something in sip.conf regarding autoregexten or something where when a SIP account registers with Asterisk it automatically adds an entry to the dialplan. If this were employed you could forward a call to all 4 Asterisk servers and only the one that had the registered user would forward the call. There are lots of ways to skin the cat but the SIP Proxy is the best way to utilise mutliple Asterisk servers when being used for SIP calls. Regards, Greyman. ___ -- 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] Polycom IP600 + PC share same switch port with VLAN
Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? 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] Polycom IP600 + PC share same switch port with VLAN
You can paste and copy nterface FastEthernet2/0/1 switchport access vlan 20 switchport mode access switchport voice vlan 120 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast - Original Message - From: Lee, John (Sydney) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN Date: Fri, 29 Feb 2008 16:39:24 +1100 Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? 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 -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.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] Problems with removing zaptel
On Fri, Feb 29, 2008 at 12:30:49AM +0100, Christian wrote: Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel modules some of them cannot be removed. The other module is in use. Also if I reboot my system they're all loaded again. Any thoughts? modprobe -r does not recursively remove modules. Try: /etc/init.d/zaptel unload #if using the init.d script from the deb /etc/init.d/zaptel stop # if using the init.d script from the tarball (The reason for the difference: there is no point in unlading a module on system shutdown. Only serves to increase the crash potential) -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with setting up Zaptel
On Thu, Feb 28, 2008 at 08:08:49PM +0100, Christian wrote: Hi all, I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just trying to get it working. But no luck as of yet. In /etc/zaptel.conf I've set the following options: fxsks=2 fxoks=1 loadzone=se defaultzone=se And in /etc/asterisk/zapata.conf I've not sure what to set exactly. For example, under [trunkgroups] what to specify there? Under [channels I set something like the following: usecallerid=yes cidsignalling=dtmf cidstart=polarity hanguponpolarityswitch=yes signalling=fxs_ks context=default Do you actually notice any problem? What is the output of: cat /proc/zaptel/* asterisk -rx 'zap show channels' You also seem to be missing some 'channel =' lines in your zapata.conf . -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN
Thanks very much for the quick response. However, switchport voice vlan.. I thought is only valid for CISCO phones and I am using Polycom and thus it would not work. Furthermore, I have already tried switchport voice vlan... before I emailed to the list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob G Sent: Friday, 29 February 2008 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN You can paste and copy innterface FastEthernet2/0/1 switchport access vlan 20 switchport mode access switchport voice vlan 120 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast - Original Message - From: Lee, John (Sydney) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN Date: Fri, 29 Feb 2008 16:39:24 +1100 Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? 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 -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.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
[asterisk-users] Can call in but cannot call out (CHANUNAVAIL): TE410 + Asterisk 1.4.13 + Zaptel 1.4.6 + Libpri 1.4.2
I encountered this strange problem which is I can call into Asterisk box but I cannot call out. I was successful before using exactly the same euroISDN line but with TE110 and different versions of Asterisk. This time, I am using: . TE410 . Asterisk 1.4.13 . Zaptel 1.4.6 . Libpri 1.4.2 1) I put the following into extensions.conf to get to the outside line exten = 0,1,Dial(Zap/1) 2) When I hit '0' on the phone, it came back like this on the console: [Feb 29 18:15:40] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring rxwink == Parsing '/etc/asterisk/users.conf': Found -- Executing [EMAIL PROTECTED]:1] Dial(SIP/5166-b7b0caa0, Zap/1) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1CLI -- Zap/1-1 is proceeding passing it to SIP/5166-b7b0caa0 -- Channel 0/1, span 1 got hangup request, cause 31 [Feb 29 18:16:12] WARNING[7365]: app_dial.c:746 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'Zap/1-1' [Feb 29 18:16:12] NOTICE[7365]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/5166-b7b0caa0' status is 'CHANUNAVAIL' 3) When I did reload chan_zap.so -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring switchtype [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring pridialplan [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring prilocaldialplan [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring overlapdial [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring priindication [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring signalling -- Reconfigured channel 1, ISDN PRI signalling [...] -- Reconfigured channel 31, ISDN PRI signalling [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring rxwink == Parsing '/etc/asterisk/users.conf': Found 3) Something dodgy when I did a # cat /proc/zaptel/1 It complains that the channels are all In Use and I am only using span 1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 ClockSour ce 1 TE4/0/1/1 Clear (In use) 2 TE4/0/1/2 Clear (In use) [...] 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use) [...] ___ -- 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
