Re: [asterisk-users] Question about voip.ms service.
Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Question about voip.ms service.
I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *naren *Sent:* Tuesday, September 13, 2011 2:22 PM *To:* John Novack *Cc:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Question about voip.ms service. ** ** Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? ** ** Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. ** ** The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. ** ** But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) ** ** Thanks ** ** On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: ** ** I also found this... seems like voip.ms outbound is broken for now! ** ** http://pbxinaflash.com/forum/showthread.php?t=10735 ** ** ** ** On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, ** ** I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? ** ** I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. ** ** I would really appreciate it if you could post the relevant section of your sip.conf for me. ** ** Thanks! Naren ** ** On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx ** ** 'slam-dunk.' ** ** Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! ** ** is ** ** Suggest you check your firewall and your configs, and above all post some more information ** ** IAX ** ** If you really want to upset some, top post as I have just done! ** ** Agreed. ** ** The real issue is communication, top bottom or in the middle ** ** Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
Re: [asterisk-users] Question about voip.ms service.
Yup, that part I got. What I am not clear about is how to set up the DID to go to my URI. When I select manage DIDs and click on the one I want to change, I see the following options for routing the DID x SIP/IAX - [main account] IAX2/10 - with my account number x SIP URI - SIP:mysi...@myuri.com:5060 x System - Hangup There are several other options but they are not selectable for me because I have not set up to use them. I used to have the routing set to SIP URI where I was able to specify my URI where the call was routed to. But with the SIP/IAX option I do not have that ability. I am missing something fundamental here. My asterisk has the iax.conf and extensions.conf entries ready to receive calls from voip.ms, but I don't know how to tel voip.ms to send the calls to my asterisk with the IAX protocol. I understand this is probably a question for the voip.ms folks, but since a couple of people mentioned earlier that they were rocking with IAX, I thought it would be an easy question for them to point me in the right direction. Thanks. On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel dai...@pervasivetelecom.comwrote: I was lurking in this conversation and I went to look more carefully at the voip.ms site. I found sample files at http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 Hope that helps. On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote: I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk
Re: [asterisk-users] Question about voip.ms service.
That's what I am hoping to do as well. Could you share some insight on how you set up the DID on the voip.ms web site to forward to Asterisk using IAX? In particular I am trying to find out where you set the url / ip address of your asterisk installation on the voip.ms web site. Thanks! On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.comwrote: I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone -- _ -- 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] Question about voip.ms service.
Ok that makes sense. I will take a look at my set up and see why it is not registering with voip.ms. I opened a ticket with voip.ms as well about an hour ago. I do like their service as well, that is why I want to try and get it working with them. Thanks John. On Tue, Sep 13, 2011 at 5:29 PM, John Novack jnov...@stromberg-carlson.orgwrote: Voip.ms has excellent support if you need it, which many do not. You log in to your account, then you can change from SIP to IAX, and if you click on the correct link they will give you your sample with your account information You need to set up a registration line in IAX, then a context in IAX that points to a context in extensions.conf Registration takes care of voip.ms finding you their web site setup is about as complete a site as I have seen, with many more options than I would ever need The only somewhat confusing issue is when using IAX they will not show you as registered Your Asterisk will, though. John Novack naren wrote: That's what I am hoping to do as well. Could you share some insight on how you set up the DID on the voip.ms web site to forward to Asterisk using IAX? In particular I am trying to find out where you set the url / ip address of your asterisk installation on the voip.ms web site. Thanks! On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.comwrote: I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Question about voip.ms service.
I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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] Question about voip.ms service.
I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Question about voip.ms service.
Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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
[asterisk-users] Re: Migration to Asterisk 1.4
Hi, I am trying to migrate my asterisk 1.0 to 1.4. I have downloaded Asterisk Now and installed. I am using the GUI that comes with Asterisk Now. I am trying to setup an internal phone system where I can make calls between extensions. I am able to create users using the interface. I am not able to figure out how to define the phones and assign them to the users. I have manually modified iax.conf and added my iaxcomm softphone and iaxy phone which seem to be registering. I am not able to figure out how to connect a user with one of these phones.I am not sure if this is the corret way to do. Any help is appreciated. Thanks, Naren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help implementing call center features of Asterisk
I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help configuring Asterisk server
I need to configure / migrate Asterisk server from 0.9 to the latest version with some upgrades. Please help! Thank you. Sincerely, Naren Koka (480) 829-0479 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Congestion error
I am using Asterisk with Connect.VoicePulse. Of late, we are getting too many congestion errors. Chris Icide has helped me before in setting up the server. He has done a wonderful job. It has worked very well until about 2 months ago. Now I need some help to fix this issue. I appreciate the help. Sincerely, Naren Koka (480) 829-0479 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Registration with servers
I have 2 servers that I use to talk from one place to another place. One of them, Server A registers with the other one, Server B. There are many cases the registration drops out and then works again after some time. The internet connection between them is not so great, which could be suspected. Server A also registers with VoicePulse. The connection to VoicePulse always works. It is only the connection with server B that fails often. Is there a timeout period that can be adjusted to maintain the connection despite bad internet? Is there a way to force the connection attempts to be more frequent? Here is the configuration on the 2 servers. iax.conf on server A ; Register with Indidge US server register = serverB:[EMAIL PROTECTED] [serverB] type=friend auth=md5 secret=password host=11.11.11.11 context=ctx disallow=all allow=ilbc qualify=yes notransfer=yes iax.conf on server B [serverA] type=friend auth=md5 secret=password host=dynamic context=ctx disallow=all allow=ilbc qualify=yes notransfer=yes If anyone could suggest some solution, it is appreciated. Thank you. Sincerely, -- Naren Koka VP of Technology INDIDGE SYSTEMS (480) 829-0479 x111 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIPURA does not register with Asterisk
I have a SIPURA 2000 which is supposed to register with the Asterisk server. However it does not register at all. I have two lines registered in the SIP.conf [201] type = friend host = dynamic secret=201 dtmfmode = rfc2833 context = default [EMAIL PROTECTED] canreinvite=no callerid = 201 201 The web interface comes up. In Line1 screen I have Register=Yes and Proxy set to the Asterisk server IP address. The Info screen shows that it has received broadcast packets but has not sent any. I did not see any registration messages on the Asterisk screen. Please help! Thanks, Naren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Support
Hi, I am looking for some help with my Asterisk system. I have a server setup with one digium card with a single connection. It is running on a 2000+ Athlon system. We are using CISCO 30 VIP and 12 SP phones. I also have a SIPURA to which I can connect analog phones. The phone system is far below expectation. The voice quality is very poor, too much echo and has issues with dialing keys - it does not recognize calling card numbers and extension numbers and I am not able to get clear voice messages. We have been using 1 line. Now I need to expand it to take 2 or 3 lines. Our current Digium card supports only one line. I need some recommendation on the hardware. I would like to see a commercial grade Asterisk installation and to find out what it takes to get one. I am looking for some help. Thanks, Naren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Eating Digits
When I call a PBX system and enter digits, Asterisk is
eating away some digits. For example when I call ATT
and when the system prompts me to enter my phone
number, Asterisk eats away some digits, so ATT does
not get the number that I entered. I am using the
extensions.conf as it came from the install with some
additions. I added longdistance to the default
context. Please help!
[default]
include = mainmenu
include = longdistance
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
Thank you,
Naren
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[Asterisk-Users] Call forwarding
I am using CISCO 30 VIP and CP 12+ IP phones. I am using 2 analog phones connected to a SIPURA. I am using chan_skinny for the CISCO phones. On the CISCO phones, only the basic phone functionality works. I can not transfer calls or anything using the chan_skinny. The analog phones also work as basic phones. From my earlier emails, I found out that chan_skinny does not support the advanced feature like this. Chan_sccp did not work with these two types of CISCO phones. I am looking for at least one phone in the system which can be the operator phone. I expect this phone to receive calls and if necessary transfer the call to an extension. Is there any possibility that I can do that with my existing phones. Otherwise, which are the recommended phones to get this functionality? Thanks, Naren __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfering
I am using CISCO 30 VIP and CP 12+ IP phones. I am using 2 analog phones connected to a SIPURA. I am using chan_skinny for the CISCO phones. On the CISCO phones, only the basic phone functionality works. I can not transfer calls or anything using the chan_skinny. The analog phones also work as basic phones. From my earlier emails, I found out that chan_skinny does not support the advanced feature like this. Chan_sccp did not work with these two types of CISCO phones. I am looking for at least one phone in the system which can be the operator phone. I expect this phone to receive calls and if necessary transfer the call to an extension. Is there any possibility that I can do that with my existing phones. Otherwise, which are the recommended phones to get this functionality? Thanks, Naren __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO 30 VIP phone / 12 SP+ Connection does not free up
Hi, I am using a 30 VIP phone and a 12 SP+ phone with Asterisk. When I complete a call outside through the ZAP device, the phone does not go back to dial tone, even after I hang up. The line gets disconnected as per Asterisk console. But the phone stays in the same state like it is connected. The ZAP line is freed up and I could make calls from other phones. Only this phone just remained in the same state. Has anyone seen this behavior? Can anyone suggest a fix to this? Thanks, Naren __ Do you Yahoo!? Yahoo! Movies - Buy advance tickets for 'Shrek 2' http://movies.yahoo.com/showtimes/movie?mid=1808405861 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO 30 VIP and 12 SP+
I am trying to connect the above 2 models to Asterisk server. The firmware seems to be 2.02 and 2.04. The 30 VIP is connecting to the network with DHCP. If anyone is using these phones, can you please give me some help. Thanks, Naren __ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
