Re: [asterisk-users] Vitelity Setup
Finally I got it working by removing the pfsense firewall. Something to do with pfsense firewall. Regards On Mon, May 28, 2012 at 2:36 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Actually I understood that register line is not required, also since my PBX is behind the pfsense firewall, now what i am going to do is putting the PBX directly in public network (i.e. without firewall) and will check whats going to happen. Hope things would sort out. Regards. On Sat, May 26, 2012 at 2:48 AM, Stephen J Alexander sjalexan...@mpbx.com wrote: If your server says it is registered, that could be part of the problem. Vitelity doesn't use trunk registration, only IP authentication. You should not be using a registration string in your trunk definition. I don't know if it will hurt but it won't help. It sounds like you might have only 1 trunk defined, but you need 2; one for inbound and one for outbound. Their servers for incoming calls and for outgoing calls are separate. If fixing that doesn't do the job, make sure that incoming traffic from Vitelity is correctly routed to your PBX (and that they have the correct IP to send SIP traffic to). Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, May 25, 2012 at 4:12 PM, Ralph Green sira...@gmail.com wrote: Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
Actually I understood that register line is not required, also since my PBX is behind the pfsense firewall, now what i am going to do is putting the PBX directly in public network (i.e. without firewall) and will check whats going to happen. Hope things would sort out. Regards. On Sat, May 26, 2012 at 2:48 AM, Stephen J Alexander sjalexan...@mpbx.comwrote: If your server says it is registered, that could be part of the problem. Vitelity doesn't use trunk registration, only IP authentication. You should not be using a registration string in your trunk definition. I don't know if it will hurt but it won't help. It sounds like you might have only 1 trunk defined, but you need 2; one for inbound and one for outbound. Their servers for incoming calls and for outgoing calls are separate. If fixing that doesn't do the job, make sure that incoming traffic from Vitelity is correctly routed to your PBX (and that they have the correct IP to send SIP traffic to). Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, May 25, 2012 at 4:12 PM, Ralph Green sira...@gmail.com wrote: Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
Is your IAX2 peer registered? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralph Green Sent: Friday, May 25, 2012 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vitelity Setup Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports - more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests - check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
If your server says it is registered, that could be part of the problem. Vitelity doesn't use trunk registration, only IP authentication. You should not be using a registration string in your trunk definition. I don't know if it will hurt but it won't help. It sounds like you might have only 1 trunk defined, but you need 2; one for inbound and one for outbound. Their servers for incoming calls and for outgoing calls are separate. If fixing that doesn't do the job, make sure that incoming traffic from Vitelity is correctly routed to your PBX (and that they have the correct IP to send SIP traffic to). Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, May 25, 2012 at 4:12 PM, Ralph Green sira...@gmail.com wrote: Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. In the Vitel web app you ust set the routing method to the IP of your pbx, maybe that's what's happening I'm pretty sure they check that the outbound calls use the same IP. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
yes I did that, even then i am not able to make outbound and inbound as well. On Thu, May 24, 2012 at 12:42 PM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. In the Vitel web app you ust set the routing method to the IP of your pbx, maybe that's what's happening I'm pretty sure they check that the outbound calls use the same IP. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). Cheers, Jeff LaCoursiere SunFone On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander wrote: Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). We had lot's of trouble with the 800 numbers as well but after help from Vitelity's support we were able to determine that the problem was that toll free require _exactly_ 10 digits to accept the toll free call. Regarding call to the Caribean we had a lot trouble with cell phones in Venezuela and it seems they were using pre-paid lines that ran out money but they eventually got around and solved it. So I think that if you insist with their support they usually resolve the issue. Best, -- Alejandro Imass Cheers, Jeff LaCoursiere SunFone On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander wrote: Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). We had lot's of trouble with the 800 numbers as well but after help from Vitelity's support we were able to determine that the problem was that toll free require _exactly_ 10 digits to accept the toll free call. That's not the trouble we were having - if you call these large companies and sit on hold waiting for an agent, then finally get transferred, you get a message Cannot complete this call from your location. Oddly, it is the same message no matter what large company you are calling. It was reproducible every time, not random. Through other carriers we work with the same numbers were no problem. I finally concluded that these large companies are probably offloading their support overseas, and that they were doing some kind of PRI transfer, offloading the new leg upstream in some manner that eventually resulted in the call being rejected. Why this seems to happen exclusively through Vitelity I can't say. Support emails went unanswered. With so many other termination providers it was easier to simply switch our 800 carrier than chase this down with a support infrastructure that won't answer emails. Regarding call to the Caribean we had a lot trouble with cell phones in Venezuela and it seems they were using pre-paid lines that ran out money but they eventually got around and solved it. So I think that if you insist with their support they usually resolve the issue. Our troubles were first with ALL calls to Jamaica. It took all day to get someone to look into it, and two days later we still couldn't complete calls there. Again, switched to a different carrier and the problem went away. Next was Trinidad. Same story. Haven't gone back to see if they were eventually resolved. I don't know who they are trying to use in the Caribbean to save cost on their routes, but I would rather work with someone that is using white routes and pay a bit more than spend all my time resolving call routing for my customers. Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? Thanks, -- Alejandro Imass Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
