[asterisk-users] Problem making international calls
Hello, I'm having problems making international calls from our asterisk using an ISDN30 in the netherlands. Below is the zapata.conf that works for all national calls, but international calls all fail a RC=41 ; zapata.conf [trunkgroups] [channels] language=nl signalling=pri_cpe switchtype=euroisdn callerid=asreceived ;pridialplan=unknown ;prilocaldialplan=unknown immediate=no callerid=asreceived ;nationalprefix=0 ;internationalprefix=00 ;faxdetect=incoming ;transfer=yes overlapdial=no group=1 context=from-pstn channel = 1-15,17-31 language=nl If I follow the info from voip-info.org I should use the parameters pridialplan and prilocaldialplan and set both to unknown for KPN in the Netherlands, however if I do so, I can't make national calls anymore, must be overlooking something here. Any help is welcome, Filip -- IT Operations Home Automation Europe Joan Muyskenweg 22 1096CJ Amsterdam KvK: 34187907 T. +31 20 4621680 D: +31 20 4621683 F. +31 84 8378748 M. +31 651 744702 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend quality wholesale termination - Singapore and Sydney, Aus
I can provide you termination to both countries, premium traffic, contact me off the list. On Mon, Jul 21, 2008 at 11:18 AM, MFH [EMAIL PROTECTED] wrote: Can anyone recommend decent quality as close to pay-as-you-go SIP wholesale termination providers in both Singapore and Sydney, Australia? I will be in both places and want a local carrier while I'm there. It needs to be easy in and easy out and if it's not $0 base or close I'll need to be able to drop it in a month. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Require Billing solution for Calling Cards retail...
I used A2billing and its pretty good. On Sat, Jul 5, 2008 at 6:07 PM, Daniel Varella [EMAIL PROTECTED] wrote: Hello Kashif, Do you have something already working ? Here in Brazil I've worked on some projects using Asterisk to make some passive call-centers receive calls from their remote customers. Is it what your customer is looking for ? About calling cards, Is something like pre-paid cards ? Do they have some system working, even without IVR ? Regards. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Linux Professional Certified LPI000143643 On Sat, Jul 5, 2008 at 9:00 AM, Kashif Naeem [EMAIL PROTECTED] wrote: Hello All, One of our French client is dealing in Wholesale termination business. Now they are going to start retail of Calling Cards. They need complete IVR and billing solution for it. Any one who has already provided such solutions please contact. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Terrible Experience Net2phone A-Z termination
I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED]wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
you can try inphonex.com Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Current available allarms in the Asterisk
Hi All; What are the current available allarms in Asterisk? In other words, based on what the allarms happen in Asterisk? Is it on the core dump or there are another factors that generate allarm? Also, where I can confirgure these allarms rules, and to where I can send the allarm (my mobile or to the email)? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] appconference low quality g729
Dear, compiling appconference 2.0. with g729 enabled, makes the quality of voices too low, for low voices , there is'nt any problem, but normal voices have alot of noises. best Mani ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP voicemail import
Hi, i've switched from the old vm-storage to imap-storage. Is there a script that can import the old messages? Regards Andreas _ Buy, rent, invest property online today. http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fwww%2Eallrealestate%2Eco%2Enz%2Freview%2Fhome%2Dbuying%2Dinfo%2Ehtml%3Frsf%3Dmsnnz%5Ftextlink_t=26000_r=REA_NZ_tagline_m=EXT___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on VMware Workstation 6
Hi, Agreed. Asterisk on a VM appears to work sometimes, only if magic is involved. It is not the way to run anything for a business. Steve On 25 Sep 2008, at 02:36, Dean Collins wrote: Mike, Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html problem solved. If you are worried about good call quality it’s either a dedicated pc or a dedicated appliance, one or the other. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Michael J. Liberatore Sent: Wednesday, 24 September 2008 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on VMware Workstation 6 Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. Its basically a iax2 connection to my voip provider and a sip connection to my phone. It does work well, but the calls especially the voicemail are all garbarled alot. Its definetly not the provider or internet connection because i use this provider for many clients asterisk setups and i also even setup a temp. asterisk setup on this very pc to test to make sure it was infact vmware causing the problem. I upgraded from vmware player to the latest vmware workstation hoping that would fix the problem since its a better system but it hasnt. I also installed and compiled the vmware tools when i installed workstation version. Is this a known issue with vmware? Is there a way to correct the issue either on the windows/vmware side or on the asterisk/linux side? Any other ways to do this project? i looked into astwind or something but either couldnt get it to work or it was unreliable. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
Bruno Castelo Branco wrote: you can try inphonex.com Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Funny Net2Phone comes up. We talked to them when we were starting out and they wanted to charge $500 setup fee because we had no volume. The guy said We have to charge this because we had many people coming to us without volume, so we charge this setup fee in order to allow us to still provide them service. Like that makes any sense to anyone. Either way, they had the worst rates in the market and claimed extremely high quality. I'm glad we didn't go with them. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
The fax is originated from a fax machine connected to an ata which supports t38. On Wed, Sep 24, 2008 at 11:54 PM, C F [EMAIL PROTECTED] wrote: On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Sorry to interrupt. I need some help regarding fax passthru mode. We are trying to configure fax passthru mode in asterisk using sip. For out of network calls/fax we use trunk configuration. i am using asterisk 1.4.2. The user has to use fax machine connected to their ata and dial the callee number, the call is originated just like a regular voice call. have not defined any special context for sending faxes. Have enabled t38 and canreinvite in peer/user and trunk configuration. But the fax is not going thru. Our service provider does support fax passthru. Following is the trunk and user/peer configuration: They support passthru, and the originating send fax is what? PSTN? or VoIP ATA with t38 support? There has to one that does the t38, if the point where it gets converted to VoIP does not support t38 then passthru will not help you. TRUNK CONF [TRUNK-OUT] type=peer host=XXX port=5060 context=default country=us dtmfmode=rfc2833 restrictcid=no canreinvite=yes insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm promiscredir=yes t38_udptl=yes USER/PEER [abc] username=abc type=friend secret=123 qualify=25000 nat=yes mailbox=12129339037 insecure=port,invite incominglimit=2 outgoinglimit=2 intl_trunk=TRUNK-OUT local_trunk=TRUNK-OUT host=dynamic dtmfmode=inband context=uscan canreinvite=yes callerid=Rizwan Qureshi 122 accountcode=1:0:abc amaflags=default disallow=all allow=ulaw allow=alaw allow=gsm t38_udptl=yes Any solutions? On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro [EMAIL PROTECTED] wrote: ATAs work OK I guess, just make sure to use a loss less codec such as ULAW. Since the OP stated he is using E1 lines then he should probably be using alaw instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
Who are you using now? We need someone that has international traffic with good rates and good quality. On Thu, Sep 25, 2008 at 7:32 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Bruno Castelo Branco wrote: you can try inphonex.com Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Funny Net2Phone comes up. We talked to them when we were starting out and they wanted to charge $500 setup fee because we had no volume. The guy said We have to charge this because we had many people coming to us without volume, so we charge this setup fee in order to allow us to still provide them service. Like that makes any sense to anyone. Either way, they had the worst rates in the market and claimed extremely high quality. I'm glad we didn't go with them. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP TLS
Hi friends, I'm doing my college final work about SIP security in Asterisk. I was reading TLS RFC and I saw that the TLS protocol has two layers that can be used in different ways. I need to know if anyone can help me with documentation or just explain with high level of details how does asterisk handle TLS protocol. I need to know too, if there's a documentation about how to configure TLS in Asterisk 1.6. That's all. -- Thank you, Rafael Puga http://whitesight.wordpress.com/ Dados olhos suficientes, todos os erros são triviais. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
I've been happy with Broadvox if you can meet their minimum requirements. I use them as one of my termination carriers for both A-Z and domestic traffic and have been happy with their quality. their rates aren't the lowest for A-Z but good quality just the same. Anyone else have a review of Broadvox to share either good or bad? Tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of broadband Voice Sent: Thursday, September 25, 2008 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Terrible Experience Net2phone A-Z termination Who are you using now? We need someone that has international traffic with good rates and good quality. On Thu, Sep 25, 2008 at 7:32 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Bruno Castelo Branco wrote: you can try inphonex.com http://inphonex.com/ Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Funny Net2Phone comes up. We talked to them when we were starting out and they wanted to charge $500 setup fee because we had no volume. The guy said We have to charge this because we had many people coming to us without volume, so we charge this setup fee in order to allow us to still provide them service. Like that makes any sense to anyone. Either way, they had the worst rates in the market and claimed extremely high quality. I'm glad we didn't go with them. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com http://www.escapetel.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
We don't do A-Z, our main source of revenue now is Cuba and a few sections of Latin America. For those routes we've found you get a better quality/price ratio by going directly to individual providers that specialize in the particular areas you're looking for. Although when we were looking around for decent a-z providers some tested out pretty well. Nergy Telecom(http://www.nergytelecom.net/) seemed to have pretty good quality for ok rates. Ipsmarx had one of the worst a-z packages I've ever seen and group3(group3.ca) has really cheap rates to a lot of destinations. The quality with them varies a lot, some routes have horrible quality but for some destinations they have really great quality for a good price. On Thu, Sep 25, 2008 at 7:43 AM, broadband Voice [EMAIL PROTECTED] wrote: Who are you using now? We need someone that has international traffic with good rates and good quality. On Thu, Sep 25, 2008 at 7:32 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Bruno Castelo Branco wrote: you can try inphonex.com Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Funny Net2Phone comes up. We talked to them when we were starting out and they wanted to charge $500 setup fee because we had no volume. The guy said We have to charge this because we had many people coming to us without volume, so we charge this setup fee in order to allow us to still provide them service. Like that makes any sense to anyone. Either way, they had the worst rates in the market and claimed extremely high quality. I'm glad we didn't go with them. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing after console dsp hangup
I have a simple context that connects to the console dsp which works, but then after I hangup I hear ringing on the console dsp. It rings until I stop asterisk. Why is that and how can I stop it? Thanks, Jerry [paging] exten = s,1,Answer exten = s,n,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup exten = h,1,Hangup -- Executing [EMAIL PROTECTED]:1] Answer(SIP/192.168.1.8-089177a8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Playback(SIP/192.168.1.8-089177a8, beep) in new stack -- SIP/192.168.1.8-089177a8 Playing 'beep' (language 'en') -- Executing [EMAIL PROTECTED]:3] Dial(SIP/192.168.1.8-089177a8, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp -- ALSA/default answered SIP/192.168.1.8-089177a8 Hangup on console == Spawn extension (paging, s, 3) exited non-zero on 'SIP/192.168.1.8-089177a8' -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/192.168.1.8-089177a8, ) in new stack == Spawn extension (paging, h, 1) exited non-zero on 'SIP/192.168.1.8-089177a8' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Do You Know What the Problem With CDMA is?
It's OT but I thought it was funny enough to point out seeing this is a telephony list.. world wide data.like the 'world series' of baseball if you ask me :-) Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile From: http://deancollinsblog.blogspot.com/2008/09/do-you-know-what-problem-wit h-cdma-is.html Subject: Do You Know What the Problem With CDMA is? I'm a Cingular customer but i noticed that the HTC Touch Pro is only available from Sprint at the moment. So when I noticed a Sprint mailer in the post box I thought I'd flick through it. Take a look at this picture and tell me if you cant work out what is wrong with CDMA http://1.bp.blogspot.com/_jmYevHrBr6M/SNqpnwbZW-I/Aww/p2pRhDO9t VM/s1600-h/Sprint.png Yep you got it world coverage includes all the countries of the world like USA, Canada and Mexico, and not something like USA and 212 other countries globally. BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it into a funding bill for the war that they had to use CDMA to 'support usa businesses'), of course that means that they now use a different handset type to all of their neighbours... though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring trunk
Hello, everyone, You Sanem which I can use software to track use of channels of ZAP asterisk. I have 4 asterisk servers with each 4E1, I would like to monitor the doors of E1, someone knows a tool for that? Thank you very much. Rodrigo Florianópolis - Brazil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
broadband Voice wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users www.voxitas.com (disclaimer, I work for this company now) -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
On September 25, 2008 09:01:52 am Dean Collins wrote: Yep you got it world coverage includes all the countries of the world like USA, Canada and Mexico, and not something like USA and 212 other countries globally. BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it into a funding bill for the war that they had to use CDMA to 'support usa businesses'), of course that means that they now use a different handset type to all of their neighbours... though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) That doesn't mean that GSM towers won't be built, it just means that the CDMA towers will be there first. I dunno; GSM 3G is all CDMA tech anyway. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
Hi. You can test Fonet Global Inc., its a good company and provide you world termination, aditional services, good rates, etc. and they work with Asterisk many years ago. www.fonetglobal.com At 06:32 a.m. 25/09/2008, Igor Hernandez wrote: Bruno Castelo Branco wrote: you can try inphonex.com Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Funny Net2Phone comes up. We talked to them when we were starting out and they wanted to charge $500 setup fee because we had no volume. The guy said We have to charge this because we had many people coming to us without volume, so we charge this setup fee in order to allow us to still provide them service. Like that makes any sense to anyone. Either way, they had the worst rates in the market and claimed extremely high quality. I'm glad we didn't go with them. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Plan Issues
Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards Tariq-- You might try a trixbox users mailing list. There might be a few trixbox users hanging around in this group who might be able to help, but your chances are much better in that list. murf AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
Andrew Kohlsmith (lists) wrote: On September 25, 2008 09:01:52 am Dean Collins wrote: Yep you got it world coverage includes all the countries of the world like USA, Canada and Mexico, and not something like USA and 212 other countries globally. BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it into a funding bill for the war that they had to use CDMA to 'support usa businesses'), of course that means that they now use a different handset type to all of their neighbours... though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) That doesn't mean that GSM towers won't be built, it just means that the CDMA towers will be there first. I dunno; GSM 3G is all CDMA tech anyway. Yes, but it is a standard agreed upon by a large number of carriers around the world. Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... regards, Drew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
On Thu, Sep 25, 2008 at 9:58 AM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On September 25, 2008 09:01:52 am Dean Collins wrote: Yep you got it world coverage includes all the countries of the world like USA, Canada and Mexico, and not something like USA and 212 other countries globally. BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it into a funding bill for the war that they had to use CDMA to 'support usa businesses'), of course that means that they now use a different handset type to all of their neighbours... though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) That doesn't mean that GSM towers won't be built, it just means that the CDMA towers will be there first. I dunno; GSM 3G is all CDMA tech anyway. -A. Dean, Quoting you: though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) I know it is an eventuality, I don't believe we are liberating only Iraq and Afghanistan (at least publically), as well as Iran, Venezuela and Peru in the near future. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
On September 25, 2008 10:41:45 am Drew Gibson wrote: Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... You don't have ATT towers near airports? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon people please post the announcement
Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
A lot of places you still can't get GSM in the US.it has improved...but GSM 3G coverage is lacking compared to EVDO/CDMA. Another option is a World Phone that can do all bands. My story; Visiting American lands in Kuala Lumpur and checks phone for messages... Puzzled look appears as it worked at Home, Canada, Mexico, Caribbean... You start to explain about GSM and their eyes open wide as they realize they need a unlocked GSM phone from a electronics shop and SIM chip from some company named Digi sold in 7-Eleven and some scratch off cards for refills using SMS. In reality my roaming fees for Intl are too high, I'll get a pre-paid in-country phone before I get phone bill for Intl roaming. My data connection syncs email all day long. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith (lists) Sent: Thursday, September 25, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is? On September 25, 2008 10:41:45 am Drew Gibson wrote: Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... You don't have ATT towers near airports? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 is asking me for Mailbox #
Joseph wrote: I just installed *-1.4 and when I enter mail extension it keep asking me for Mailbox # I have in sip.conf under my extension mailbox=11 type=friend *-1.2 was jumping straight to messages. What did change? When you call VoiceMailMain, you need to provide the mailbox number as an argument to the application if you don't want to be prompted for a mailbox number. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
http://bit.ly/asterskype ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
I know a lot of linux and open source people think it's superfluous, but a pseudo chan_skype is huge (assuming it works as advertised). It means anyone with Skype can connect to your server presence. And presumably you can call people via Skype. And use Skype out, etc. On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness [EMAIL PROTECTED] wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype-asterisk connection announced (was Astricon people please post the announcement)
This has been talked about for at least two years. I know Mark was into it, but had been silent about it for many moons. Now it appears to become a reality. If I had a running asterisk instance I'd jump into the beta right now. We'll be gabbing about this tomorrow on the VUC: http//voipusersconference.org Astricon recap edition. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
It's essentially a channel driver. Licensed per channel in the same way that the g729 codec is. Limited private beta opening soon. Tim. On 25 Sep 2008, at 17:47, Steve Anness wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
They demoed it - everyone seems pretty confident it works as advertized. No wide-band codec (yet) Tim. On 25 Sep 2008, at 17:55, randulo wrote: I know a lot of linux and open source people think it's superfluous, but a pseudo chan_skype is huge (assuming it works as advertised). It means anyone with Skype can connect to your server presence. And presumably you can call people via Skype. And use Skype out, etc. On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness [EMAIL PROTECTED] wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
Andrew Kohlsmith (lists) wrote: On September 25, 2008 10:41:45 am Drew Gibson wrote: Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... You don't have ATT towers near airports? -A Nope, no ATT north of Buffalo. To be honest, it happened a few years ago (~2002). We now have Rogers' towers near airports (and 3G iPhones in stores). Bell Canada and Telus are moving to GSM 3G (side-stepping standard GSM so they don't have to admit their mistakes) regards, Drew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI TE110P Configuration
I was configuring asterisk with TE110P Card.When run zttool It is showing Blue Alarm/Yellow Alarm/Recovering and the card's LED is blinking RED and GREEN. I have connected 12,45 Lines from ISDN modem(RAD ASMi-52) to 12,45 of the PRI card respectively. I am using Airtel's(India) ISDN connection Plese Help me to sort it out.. -- Regards, Shyju ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
Hi all, Voiceroute is twittering abt it http://twitter.com/voiceroute Video with mark on announcement will be uploaded in 1 hour. http://youtube.com/voiceroute Ming On 9/25/08, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from Gmail for mobile | mobile.google.com Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona http://www.astricon.net/2008/glendale/web/confTracks.php#t193 Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail Pavilion, Javits Center, NYC http://druidweb20.eventbrite.com DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform) http://www.voiceroute.org/druidcon VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com UC 2.0 Video - Mozilla Ubiquity + Druid http://www.youtube.com/watch?v=f-5rDBPuGRc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
On Thu, Sep 25, 2008 at 06:38:24PM +0200, randulo wrote: So Skype finally will talk to Asterisk Excellent news! Great news! You mean that there is finally a free implementation of the skype protocol so I can start using it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote: Great news! You mean that there is finally a free implementation of the skype protocol so I can start using it? Free? AFAICT, not. Neither free as in beer nor speech. Move along, nothing to see here. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 is asking me for Mailbox #
On 09/25/08 11:34, Mark Michelson wrote:
Joseph wrote:
I just installed *-1.4 and when I enter mail extension it keep asking me for
Mailbox #
I have in sip.conf under my extension mailbox=11 type=friend
*-1.2 was jumping straight to messages.
What did change?
When you call VoiceMailMain, you need to provide the mailbox number as an
argument to the application if you don't want to be prompted for a mailbox
number.
Mark Michelson
Thanks I got this one:
in *-1.2 it I had {CALLERIDNUM}
in *-1.4 it changed to {CALLERID(num)}
--
#Joseph
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Re: [asterisk-users] PRI TE110P Configuration
On 25 Sep 2008, at 18:38, Shyju K wrote: I was configuring asterisk with TE110P Card.When run zttool It is showing Blue Alarm/Yellow Alarm/Recovering and the card's LED is blinking RED and GREEN. I have connected 12,45 Lines from ISDN modem(RAD ASMi-52) to 12,45 of the PRI card respectively. I am using Airtel's(India) ISDN connection Plese Help me to sort it out.. Config? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve On 9/25/08 12:53 PM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote: Great news! You mean that there is finally a free implementation of the skype protocol so I can start using it? Free? AFAICT, not. Neither free as in beer nor speech. Move along, nothing to see here. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip forking needed for ekiga 3.0
So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu 1500 qdisc pfifo_fast qlen 1000 link/ether xx:xx:xx:xx:xx:xx brd ff:ff:ff:ff:ff:ff inet 10.75.22.1/24 brd 10.75.22.255 scope global eth0 inet 10.75.22.101/24 brd 10.75.22.255 scope global secondary eth0:1 So when ekiga (3.0) tries to place a call through Asterisk it in fact does parallel requests from all addresses. This is what appears to confuse Asterisk. Please see the above tickets for more details. Thots? b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-app_nv_faxdetect - Gentoo ebuild for *-1.4 was: NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6
On 09/25/08 08:11, Tzafrir Cohen wrote: On Wed, Sep 24, 2008 at 10:25:45PM -0600, Joseph wrote: My problme is that few lines in a source code needs to be modified before compiling it. Changing the source code is a simple thing but now the ebuild needs to be modified as well to point to the source code; too many problems. Asterisk 1.2 - 1.4 is a change in the build system. Most of it (except menuselect) is for the better). Adjusting your build scripts for that (and a packaging system is essentially a glorified build script) only takes some work. I would appreciate it if you hadn't kept your patches for yourselves. This would have also saved you some time on the next release (there are already RCs of 1.6.0 for yor test-building pleassure). BTW: maybe you need a newer version of nvfaxdetect? There has been one released, IIRC. If not, there should be such a version on agx's modules addons collection. Again, keeping your changes to yourself is bad. Also recall that for 1.4 and above you must define AST_MODULE. If you don't do so, you get very strange errors. I didn't intent to keep it for myself. I would be willing to work on it but I might needs some help as I'm not a pro programmer. If anybody is willing to help write and ebuild for asterisk-app_nv_faxdetect. We can post it on portage/overlay. Please drop me private email. -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
My thoughts are that to do parallel requests from every IP address on the machine is extremely weird behaviour. How would any server know which to respond to? SIP forking is supposed to send requests to multiple different destinations (or fork mid-stream to send to different destinations). Sending from multiple different points of origin doesn't make any sense at all in either a logical or rational fashion. What's it supposed to accomplish? N. Brian J. Murrell wrote: So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu 1500 qdisc pfifo_fast qlen 1000 link/ether xx:xx:xx:xx:xx:xx brd ff:ff:ff:ff:ff:ff inet 10.75.22.1/24 brd 10.75.22.255 scope global eth0 inet 10.75.22.101/24 brd 10.75.22.255 scope global secondary eth0:1 So when ekiga (3.0) tries to place a call through Asterisk it in fact does parallel requests from all addresses. This is what appears to confuse Asterisk. Please see the above tickets for more details. Thots? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
On Thu, 2008-09-25 at 14:56 -0400, SIP wrote: Sending from multiple different points of origin doesn't make any sense at all in either a logical or rational fashion. What's it supposed to accomplish? It seems to be a shot-gun approach to making a SIP connection. The assumption being I suppose that one or more of the IP aliases will fail for whatever reason (policy routing, filtering, etc.), so just try them all, and use the first one to make a completion and drop the others. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Dimensioning
All, I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium TE410P card, calls will come in TDM and go out VoIP, we will require to compress them using G729. What specs do I need to support for 4 E-1's with cdr logging to mysql? We're thinking about getting two servers 4 E-1's each, is it possible to fit both cards in one machine? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
That strikes me as being careless and unreliable. Call me a purist, but I'm of the opinion that you should KNOW which interface to use based on which interface is registered and choose ONE interface based on the rules you've established during registration. What happens if you want to ensure that data goes across a VPN (in order to encrypt your VoIP communications) instead of the public internet? Or if you want to ensure a particular route based on why you created your multiple interfaces in the first place? That takes all the logic out of the equation and just says, Here's a bunch of packets. Figure out what to do with them. I'll be waiting for your response. There's a reason routing rules exist and mature services allow you to control the interface from which it originates. N. Brian J. Murrell wrote: On Thu, 2008-09-25 at 14:56 -0400, SIP wrote: Sending from multiple different points of origin doesn't make any sense at all in either a logical or rational fashion. What's it supposed to accomplish? It seems to be a shot-gun approach to making a SIP connection. The assumption being I suppose that one or more of the IP aliases will fail for whatever reason (policy routing, filtering, etc.), so just try them all, and use the first one to make a completion and drop the others. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
If I am right I think you will find that you will not have enough power to run 4e1 with g729 codec on little 1950.. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, September 26, 2008 3:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Server Dimensioning All, I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium TE410P card, calls will come in TDM and go out VoIP, we will require to compress them using G729. What specs do I need to support for 4 E-1's with cdr logging to mysql? We're thinking about getting two servers 4 E-1's each, is it possible to fit both cards in one machine? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
On Thu, 2008-09-25 at 15:31 -0400, SIP wrote: That strikes me as being careless and unreliable. That's one argument. I can also see the ekiga developers' argument though and that's to strive for the most automatic functionality possible. The less things you have to ask users, the more likely you are to just work. Call me a purist, but I'm of the opinion that you should KNOW which interface to use based on which interface is registered We are talking about IP aliases here, not real interfaces. and choose ONE interface based on the rules you've established during registration. What rules would you establish during registration? What happens if you want to ensure that data goes across a VPN (in order to encrypt your VoIP communications) instead of the public internet? Presumably you have some [policy] routing that ensures that. But on the other hand, if you did have two addresses on an interface, one for VPN and one for everything else, unless you shotgun out you need to either know which address to use or ask the user. Either case may fail. That takes all the logic out of the equation and just says, Here's a bunch of packets. Figure out what to do with them. I'll be waiting for your response. I don't think it's quite that bad. It's more like here's a bunch of session requests, please complete them [you don't know it yet, but I'm going to tear down all but the first one you complete]. But the glitch is that even though I send you 3 of them, due to [policy] routing and firewalling, you might only get one. There's a reason routing rules exist and mature services allow you to control the interface from which it originates. Really, I'm just the messenger here. I doubt the ekiga team and the asterisk team would be willing to sit down and discuss who is right here, so I'm trying to be the conduit. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Create virtual extension
Have, i want to create a sip extension to a context in my dialplan. how i can do that? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
Jon Weisman schrieb: I'm planning on getting a Dell PowerEdge 1950. All I can tell is that I have bad experiences with those Dell PowerEdges. A standard Debian Etch install (2.6.18 kernel I think) didn't even have the driver to run the network interface. At least Dell doesn't seem to play nice with Debian. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
Philipp Kempgen schrieb: Jon Weisman schrieb: I'm planning on getting a Dell PowerEdge 1950. All I can tell is that I have bad experiences with those Dell PowerEdges. A standard Debian Etch install (2.6.18 kernel I think) didn't even have the driver to run the network interface. But afaicr that was a PowerEdge 2950 or something. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype + Asterisk Interview at Astricon
Hi all Interview with Mark Spencer Wilhelm Lundborg (Manager, Skype for Business) on Skype + Asterisk announcement at Astricon 2008 See video here http://www.youtube.com/watch?v=ABYkNUuShpY Follow up on coverage of Astricon Druid http://twitter.com/voiceroute Ming -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform) http://www.voiceroute.org/druidcon VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com Voiceroute videos on Druid, Open Source Unified Communications Asterisk http://youtube.com/voiceroute ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create virtual extension
Manolet Gmail wrote: Have, i want to create a sip extension to a context in my dialplan. how i can do that? ___ Simple. Use a Goto: [context1] exten = 123,1,Goto (context2,456,1) [context2] exten = 456,1,Background(tt-monkeys) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
You need to define what you mean by SIP forking. There are many things people mean by that. They are usually one of: 1) Call branching (proxies do this). 2) Parallel but distinct call legs managed by a UAC (this is what Asterisk does when you Dial(SIP/exten1SIP/exten2SIP/exten3,...)). -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mysql Command and number rows returned
Without issuing a separate loop thru a result set. Is there any way
anyone knows of to output the number of rows a mysql query returned.
Aka
..
exten = 1,n,MYSQL(Query resultid ${connid} SELECT\ `State`\ FROM\
`AreaCodes`\ WHERE\ `AreaCode`=\'${CIDArea}\')
exten = 1,n(fetchrow),MYSQL(Fetch foundRow ${resultid} number) ; fetch row
exten = 1,n,GotoIf($[${foundRow} = 1]?done) ; leave loop if no row
found
exten = 1,n,Set(State=${State})
exten = 1,n,Goto(fetchrow) ; continue loop if row found
exten = 1,n,Set(RowsReturned=MYSQL(Fetch rowcount ${resultid})
exten = 1,n(done),MYSQL(Clear ${resultid})
..
David Murphy Systems Adminsitrator
myLogo
Email: AIM:
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
lgdavidmurphy
image001.jpg___
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[asterisk-users] users.conf behavior
I have an Asterisk server running 1.4.20 and I have all my users in users.conf. Inside users.conf I used... #include ww-users.conf Thats seems to work great with one exception... The exception is that anytime anyone updates their voicemail password, Asterisk rewrites users.conf combines ww-users.conf and it removes my include line from users.conf. Is that expected behavior? I guess that I would have expected it to know to write the changes to the corresponding include file. Is there a better place to put the include? Maybe a better way to handle breaking my users up by location? Should I be using and include in the users.conf Thanks, Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
Philipp Kempgen wrote: Jon Weisman schrieb: I'm planning on getting a Dell PowerEdge 1950. All I can tell is that I have bad experiences with those Dell PowerEdges. A standard Debian Etch install (2.6.18 kernel I think) didn't even have the driver to run the network interface. At least Dell doesn't seem to play nice with Debian. I have not had this problem with Dell PowerEdge 2650s 2850s but I cannot speak to the 1950. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
We are using few dell 1950, it been two year and never had any issue, Jai www.didforsale.com *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Thu, Sep 25, 2008 at 3:19 PM, Alex Balashov [EMAIL PROTECTED]wrote: Philipp Kempgen wrote: Jon Weisman schrieb: I'm planning on getting a Dell PowerEdge 1950. All I can tell is that I have bad experiences with those Dell PowerEdges. A standard Debian Etch install (2.6.18 kernel I think) didn't even have the driver to run the network interface. At least Dell doesn't seem to play nice with Debian. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Dimensioning
Jon Weisman wrote: All, I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium TE410P card, calls will come in TDM and go out VoIP, we will require to compress them using G729. What specs do I need to support for 4 E-1's with cdr logging to mysql? We're thinking about getting two servers 4 E-1's each, is it possible to fit both cards in one machine? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'd be guessing but I don't think you'll manage more than 70 channels on it. We are running dual clovertown systems(2.33ghz) and I don't think I would want to throw more than 190 channels on it transcoding to g729. -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
Alex Balashov wrote: You need to define what you mean by SIP forking. There are many things people mean by that. They are usually one of: 1) Call branching (proxies do this). 2) Parallel but distinct call legs managed by a UAC (this is what Asterisk does when you Dial(SIP/exten1SIP/exten2SIP/exten3,...)). Exactly. These are all endpoint or middlepoint things. SIP forking is never an original starting point thing. That's just WEIRD. You fork to hit multiple endpoints simultaneously. Not one endpoint from multiple starting points. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
How is this exactly news? Hasn't chan_skype been around and available for a while now? How is this different? Eric On Thu, Sep 25, 2008 at 9:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
On Sep 25, 2008, at 11:06 AM, Steve Anness wrote: So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. The license for Skype will be the same way you license g.729. So yes, it's not free... but you're only paying for in use channel capabilities... but think of the benefits... Skype will work just like let's say SIP or ZAP or IAX2 would in the dial plan... so you could have SKYPE\username registered as an extension... or ... even in a queue which I am really excited about. You could registered as many usernames as you want... and then have as many simultaneous calls as licenses... great for calling out and even really awesome for calling in. Plus, unlike regular skype, on the callin, you can have multiple channels. It's really very exciting. Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com Using VoIP? SIP:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing after console dsp hangup
Jerry Geis wrote:
Why is that and how can I stop it?
I've never tried paging directly to the console, since it can introduce
too much feedback. Try recording the page and then play it back to the
console:
exten = s,1,Set(active=${DB(paging/active)})
exten = s,n,GotoIf($[${active} = YES]?7:4)
;
;* Set database entry for
;* paging active to YES
;
exten = s,n,Set(DB(paging/active)=YES)
;*
;* If paging currently in use,
;* jump to paging-inuse
;* context.
;*
exten = s,n,Goto(paging-inuse,s,1)
;**
;* Start recording to paging.gsm,
;* no longer then 30 seconds if
;* silence for 5 seconds, terminate
;* recording
;***
exten = s,n,Record(paging:gsm|5|30)
exten = s,n,Hangup()
;*
;* On hangup from paging, Playback paging file
;* then set paging/active to NO.
;*
exten = h,1,Dial(Console/dsp)
exten = h,n,Playback(paging)
exten = h,n,Set(DB(paging/active)=NO)
[paging-inuse]
exten = s,1,Congestion
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] How to send indicating call privacy using P-Asserted-Identity?
Or any idea how to insert Privacy header into your dial command? Zeeshan On Tue, Sep 23, 2008 at 7:00 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi, I know how to use indicating P-Asserted-Identity, but the SIP trunk provider requires to send call privacy using P-Asserted-Identity or Remote-Party-Id header. What I am doing is exten = _.,n,SipAddHeader(P-Asserted-Identity: name sip:[EMAIL PROTECTED]) The provider gets this as anonymous and can't flag the call as private. On asking them again, they say send us Invites that either contain Remote-Party-ID or P-Asserted-Identity with the correct headers flagged. Now I don't know what exactly this means and how do I do this. They are SIP trunk providers but don't deal is Asterisk. Any ideas? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Calls getting stuck in there
Did you ever solve this? I am experiencing the same issue with 1.4.21.2.
I have turned autofill on and tried incoming limits, but no luck. It
happens at least once per day. Agents will be available but calls will
just sit there until one of the waiting agents logs off and back in.
Andrew
Tariq .. wrote:
the Autofill thing didn't solve the problem.. i have another server
hosted in the USA with Asterisk 1.4.20-1 on it.. it doesn't have that
problem..
in the server i'm talking about the only way i found to avoid this
problem is to set a time out for the queue then the user is rotated into
the same queue again.. that will give the waiting users a chance to go
delivered.. i'm already questioning my agents about the delay in
answering the calls so i set the time out to 3 seconds where the caller
will be rotated in turns and the queue will be working fine..
the Autofill worked with this slution pretty well .. plus when the stuck
caller gets rotated his chance of getting connected to an agent go
higher as he won't be stuck forever..
but i need a better solution for this problem.. so im thinking of
installing the Asterisk 1.4.20-1 which i haven't faced any problem with
since i installed it.
Regards
http://www.tareksawah.com/
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sat, 13 Sep 2008 22:00:14 -0700
Subject: Re: [asterisk-users] Queue Calls getting stuck in there
Try the autofill=yes setting available in queues.conf
Original Message
Subject: [asterisk-users] Queue Calls getting stuck in there
From: Tariq .. [EMAIL PROTECTED]
Date: Sat, September 13, 2008 5:53 pm
To: Asterisk Users asterisk-users@lists.digium.com
Greetings,
i have a problem with my asterisk ..
i'm using Asterisk 1.4.19-1 with FreePBX 2.4.1.1 and TrixBox
the problem is that i'm having is the following.. a call comes to a
Queue.. the caller must be forwarded to one of the free members who are
waiting.. but instead of going to a member.. the caller stays in the
queue without being forwarded..
i tried to play with the timeout and fail over times but the caller
stays in the queue no matter what..
following are my Queues.conf , Extensions.conf, SIP.conf for one of my
queues
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[asterisk-users] Sip reload casuing issues
Howdy, Running asterisk 1.4.13 Sometime when running a sip reload the clients are unable to make and receive calls.. Any pointers? No errors in debug or asterisk console so far.. Cheers, Taff.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold for sub tenants
Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring simul calls
Howdy, Running asterisk 1.4 Is there a way to check the simultaneous sip calls in asterisk and display with mrtg or some graphing app??? Also is there a way to segregate these based on extension or context? Cheers, Taff.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAP not answering call
Hi, I am trying to interface our old PBX(Siemens) to asterisk via some analogue ZAP lines. The problem is that Asterisk never successfully answers the call. See debug ouput below. If I connect FXO - FXS on the same card and make a call it all works fine. So the card is not faulty. I see there are some stange (to me) messages in the debug. I have done search on google and tried all suggestions but they do not fix. eg. busydetect=no callprogress=no hanguponpolarityswitch=yes Some people suggest its to do with callerID. (How can I tell if the old PBX sends callerID?) have tried turning callerid in asterisk on/off - changing settings. etc. callerid=yes/no cidstart=ring/palarity cidsignalling=v23/bell/dtmf Does anyone have any other ideas? [Sep 26 12:31:54] VERBOSE[7905] logger.c: -- Starting simple switch on 'Zap/53-1' [Sep 26 12:31:54] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)... [Sep 26 12:31:55] NOTICE[7905] chan_zap.c: Got event 2 (Ring/Answered)... [Sep 26 12:31:57] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)... [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1] Goto(Zap/53-1, ivr-2|s|1) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Goto (ivr-2,s,1) [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1] Set(Zap/53-1, LOOPCOUNT=0) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:2] GotoIf(Zap/53-1, 0?begin) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:3] Ringing(Zap/53-1, ) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:4] Answer(Zap/53-1, ) in new stack [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Took Zap/53-1 off hook [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: No echo training requested [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:5] Wait(Zap/53-1, 1) in new stack *[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6* *[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669362190 [Sep 26 12:31:57] WARNING[7905] chan_zap.c: Ring/Off-hook in strange state 6 on channel 53 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669361310* [Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:6] BackGround(Zap/53-1, ssa/welcome) in new stack [Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Zap/53-1 Playing 'ssa/welcome' (language 'en') *[Sep 26 12:31:58] WARNING[7905] chan_zap.c: Ring/Off-hook in strange state 6 on channel 53* [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:7] Set(Zap/53-1, TIMEOUT(digit)=3) in new stack [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Digit timeout set to 3 [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:8] Set(Zap/53-1, TIMEOUT(response)=5) in new stack [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Response timeout set to 5 [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:9] WaitExten(Zap/53-1, |) in new stack [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669359254 [Sep 26 12:32:04] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:32:04] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:32:04] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669355306 [Sep 26 12:32:05] VERBOSE[7905] logger.c: -- Timeout on Zap/53-1, going to 't' zapata.conf - snippet [channels] ;busydetect=no ;callprogress=no ;hanguponpolarityswitch=yes ;;; line=53 WCTDM/0/21 FXSKS signalling=fxs_ks callerid=yes cidsignalling=v23 cidstart=polarity group=0 context=from-pstn channel = 53 context=default zaptel.conf # Global data loadzone= au defaultzone = au # Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) B8ZS/ESF RED span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17,31 dchan=16 # Span 2: WCTDM/0 Wildcard
Re: [asterisk-users] Music on hold for sub tenants
Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
On Thu, 2008-09-25 at 17:25 -0700, Fred Posner wrote: I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. It's still early, but still, nobody has answered my question as to whether Skype will be using my Asterisk server's CPU and bandwidth to bridge calls between anonymous third parties (i.e. two people not involved in my call plan in any way, just using my Asterisk server as a bridge for their lame, NATted connectivity) they way they do with their client. Y'all do realize with Skype that they bridge calls between two parties using a third, anonymous, (donor) party, when the two parties cannot connect to each other because their NAT and/or firewalls are too restrictive to allow them to connect directly, right? b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold for sub tenants
Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: Thursday, 25 September 2008 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Astricon people please post the announcement On Thu, 2008-09-25 at 17:25 -0700, Fred Posner wrote: I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. It's still early, but still, nobody has answered my question as to whether Skype will be using my Asterisk server's CPU and bandwidth to bridge calls between anonymous third parties (i.e. two people not involved in my call plan in any way, just using my Asterisk server as a bridge for their lame, NATted connectivity) they way they do with their client. Y'all do realize with Skype that they bridge calls between two parties using a third, anonymous, (donor) party, when the two parties cannot connect to each other because their NAT and/or firewalls are too restrictive to allow them to connect directly, right? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold for sub tenants
...since everyone else top posted. Take a look at the application setmusiconhold. CLI core show application SetMusicOnHold You can use this in a dialplan as follows: [tenant1incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tenant1sounds/welcome) exten = s,n,SetMusicOnHold(tenant1) [tenant2incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tentant2sounds/welcome) exten = s,n,SetMusicOnHold(tenant2) Use that with the previously supplied info. Darrick carl Lougher wrote: Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Lots of questions about this one. There's definitely a demand for it so I can see why Digium would be interested in exploring this option. Time will tell how well it will work. I'm personally not too excited about bolt-on binaries which are probably not compatible with uClibc (and therefore Astlinux). That leaves us in the same place as we are with codec_g729. We're at the mercy of whoever creates these binaries to produce one that will work for us. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring simul calls
Check our howto: http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cac ti-for-pretty-graphs/ and for nagios monitoring http://www.voipphreak.ca/2008/06/19/monitoring-asterisk-with-snmp-nagios-and -nagios-administrator-using-ubuntu-lts-804-server/ Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of carl Lougher Sent: Thursday, September 25, 2008 11:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Monitoring simul calls Howdy, Running asterisk 1.4 Is there a way to check the simultaneous sip calls in asterisk and display with mrtg or some graphing app??? Also is there a way to segregate these based on extension or context? Cheers, Taff.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP not answering call
Just to check - have you got the right modules plugged into the right sort of lines? Also - some analog phone interfaces are NOT standard. :( But the line modules have to be (to work with standard phone lines, of course) PaulH Daniel Johnson wrote: Hi, I am trying to interface our old PBX(Siemens) to asterisk via some analogue ZAP lines. The problem is that Asterisk never successfully answers the call. See debug ouput below. If I connect FXO - FXS on the same card and make a call it all works fine. So the card is not faulty. I see there are some stange (to me) messages in the debug. I have done search on google and tried all suggestions but they do not fix. eg. busydetect=no callprogress=no hanguponpolarityswitch=yes Some people suggest its to do with callerID. (How can I tell if the old PBX sends callerID?) have tried turning callerid in asterisk on/off - changing settings. etc. callerid=yes/no cidstart=ring/palarity cidsignalling=v23/bell/dtmf Does anyone have any other ideas? [Sep 26 12:31:54] VERBOSE[7905] logger.c: -- Starting simple switch on 'Zap/53-1' [Sep 26 12:31:54] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)... [Sep 26 12:31:55] NOTICE[7905] chan_zap.c: Got event 2 (Ring/Answered)... [Sep 26 12:31:57] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)... [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1] Goto(Zap/53-1, ivr-2|s|1) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Goto (ivr-2,s,1) [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1] Set(Zap/53-1, LOOPCOUNT=0) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:2] GotoIf(Zap/53-1, 0?begin) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:3] Ringing(Zap/53-1, ) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:4] Answer(Zap/53-1, ) in new stack [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Took Zap/53-1 off hook [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: No echo training requested [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:5] Wait(Zap/53-1, 1) in new stack *[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6* *[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669362190 [Sep 26 12:31:57] WARNING[7905] chan_zap.c: Ring/Off-hook in strange state 6 on channel 53 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669361310* [Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:6] BackGround(Zap/53-1, ssa/welcome) in new stack [Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Zap/53-1 Playing 'ssa/welcome' (language 'en') *[Sep 26 12:31:58] WARNING[7905] chan_zap.c: Ring/Off-hook in strange state 6 on channel 53* [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:7] Set(Zap/53-1, TIMEOUT(digit)=3) in new stack [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Digit timeout set to 3 [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:8] Set(Zap/53-1, TIMEOUT(response)=5) in new stack [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Response timeout set to 5 [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:9] WaitExten(Zap/53-1, |) in new stack [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669359254 [Sep 26 12:32:04] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:32:04] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:32:04] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669355306 [Sep 26 12:32:05] VERBOSE[7905] logger.c: -- Timeout on Zap/53-1, going to 't' zapata.conf - snippet [channels] ;busydetect=no ;callprogress=no ;hanguponpolarityswitch=yes ;;; line=53 WCTDM/0/21 FXSKS signalling=fxs_ks callerid=yes cidsignalling=v23 cidstart=polarity
