Re: [Asterisk-Users] Sip Trunking
On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote: > I must use sip, cos we'll use cisco rtp header-compression to save > bandwidth. > > Could you tell me the best way to send calls from asterisk1 to > asterisk2, since I cannot use IAX trunking? Maybe I'm way off base here, but I'm pretty sure that IAX2 trunking will save you more bandwidth than rtp header compression, at least if you've got multiple calls going between the two servers... (Then again, I might be a little biased, since IAX2 trunking was my idea.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
Why not use IAX2 trunking you can accomplish the same results with .. I tried SIP to SIP with asterisk you must do it without passwords. bkw On Mon, 5 Jan 2004, Eduardo Goncalves wrote: > Hi list, > > I have to connect two asterisk box, in this scenario: > > [asterisk1]sip[asterisk2]PSTN > > I must use sip, cos we'll use cisco rtp header-compression to save > bandwidth. > > Could you tell me the best way to send calls from asterisk1 to > asterisk2, since I cannot use IAX trunking? > > Thanks in advance > Eduardo > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 2004-01-05 at 10:24, Eduardo Goncalves wrote: > Hi list, > > I have to connect two asterisk box, in this scenario: > > [asterisk1]sip[asterisk2]PSTN > > I must use sip, cos we'll use cisco rtp header-compression to save > bandwidth. Will rtp header compression necessarily be better than full removal of IP headers? I'm betting rtp headers are already quite small and any compression there is minimal with respect to how IAX trunking combines what would otherwise be many packets into a single packet thus removing about 40 bytes per packet per call over 1. > Could you tell me the best way to send calls from asterisk1 to > asterisk2, since I cannot use IAX trunking? Sounds like you have already made decision to go down a path. Are you truely willing to accept better approaches? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 05 Jan 2004 10:19:24 -0700 Jared Smith <[EMAIL PROTECTED]> wrote: > On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote: > > I must use sip, cos we'll use cisco rtp header-compression to > > save > > bandwidth. > > > > Could you tell me the best way to send calls from asterisk1 to > > asterisk2, since I cannot use IAX trunking? > > > Maybe I'm way off base here, but I'm pretty sure that IAX2 trunking > will save you more bandwidth than rtp header compression, at least if > you've got multiple calls going between the two servers... I don't think it's the case. I'll have only 4 channels. On my lab tests, SIP with gsm uses 26kB/s, since the link is a frame-relay and cisco routers, I've used cisco rtp header compression, and got 16kB/s per channel. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 5 Jan 2004 11:20:08 -0600 (CST) Brian West <[EMAIL PROTECTED]> wrote: > Why not use IAX2 trunking you can accomplish the same results with .. > I tried SIP to SIP with asterisk you must do it without passwords. Because cisco doesn't compress IAX headers, only rtp. [ ]'s Eduardo > On Mon, 5 Jan 2004, Eduardo Goncalves wrote: > > > Hi list, > > > > I have to connect two asterisk box, in this scenario: > > > > [asterisk1]sip[asterisk2]PSTN > > > > I must use sip, cos we'll use cisco rtp header-compression to > > save > > bandwidth. > > > > Could you tell me the best way to send calls from asterisk1 to > > asterisk2, since I cannot use IAX trunking? > > > > Thanks in advance > > Eduardo > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 2004-01-05 at 12:47, Eduardo Goncalves wrote: > On Mon, 05 Jan 2004 10:19:24 -0700 > Jared Smith <[EMAIL PROTECTED]> wrote: > > > On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote: > > > I must use sip, cos we'll use cisco rtp header-compression to > > > save > > > bandwidth. > > > > > > Could you tell me the best way to send calls from asterisk1 to > > > asterisk2, since I cannot use IAX trunking? > > > > > > Maybe I'm way off base here, but I'm pretty sure that IAX2 trunking > > will save you more bandwidth than rtp header compression, at least if > > you've got multiple calls going between the two servers... > > I don't think it's the case. I'll have only 4 channels. > > On my lab tests, SIP with gsm uses 26kB/s, since the link is a > frame-relay and cisco routers, I've used cisco rtp header compression, > and got 16kB/s per channel. Something sounds fishy here. Asterisk sends out 50 packets a second of audio(20ms). If your numbers above are per channel, you achieved a 10k reduction in 50 packets, or 204.8 bytes average per packet. Since a GSM audio packet contains 33 bytes of audio, this large header compression sounds fishy. If you are talking bits, not bytes, then it isn't that impressive. You still will probably find more efficiency in IAX. Try it and tell us your results before shooting it down. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 2004-01-05 at 12:57, Eduardo Goncalves wrote: > On Mon, 5 Jan 2004 11:20:08 -0600 (CST) > Brian West <[EMAIL PROTECTED]> wrote: > > > Why not use IAX2 trunking you can accomplish the same results with .. > > I tried SIP to SIP with asterisk you must do it without passwords. > > Because cisco doesn't compress IAX headers, only rtp. Have you verified that IAX2 trunking doesn't save you the same or more bandwidth, or are you stuck on your solution of using the Ciscos? I think IAX2 is already a small header, and trunking saves you 8 bytes per packet per call over 1. So at 4 calls you have 32 bytes * 50 per second of saving. Thats 1800 bytes per second. > > On Mon, 5 Jan 2004, Eduardo Goncalves wrote: > > > > > Hi list, > > > > > > I have to connect two asterisk box, in this scenario: > > > > > > [asterisk1]sip[asterisk2]PSTN > > > > > > I must use sip, cos we'll use cisco rtp header-compression to > > > save > > > bandwidth. > > > > > > Could you tell me the best way to send calls from asterisk1 to > > > asterisk2, since I cannot use IAX trunking? > > > > > > Thanks in advance > > > Eduardo -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 05 Jan 2004 15:42:25 -0600 Steven Critchfield <[EMAIL PROTECTED]> wrote: > > On my lab tests, SIP with gsm uses 26kB/s, since the link is a > > frame-relay and cisco routers, I've used cisco rtp header > > compression, and got 16kB/s per channel. > > Something sounds fishy here. > > Asterisk sends out 50 packets a second of audio(20ms). If your numbers > above are per channel, you achieved a 10k reduction in 50 packets, or > 204.8 bytes average per packet. Since a GSM audio packet contains 33 > bytes of audio, this large header compression sounds fishy. If you are > talking bits, not bytes, then it isn't that impressive. You still will > probably find more efficiency in IAX. Try it and tell us your results > before shooting it down. Sorry, the results are in bits per second, not bytes. my mistake. I'm doing measure tests with SIP and IAX2 trunking. I'll finish today and post the results. Thanks for the tips -- Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunking
Brent Davidson <[EMAIL PROTECTED]> writes: > I have several branch offices, each with their own Asterisk server > (version 1.4.22.1) handling their PBX functions. All of these offices > need to talk to each other. In sip.conf I created a peer entry for each > office with a username of branch-user and a friend entry for every > branch-user with the username being just the branch, for example: You should only need peer entries... type=user is dying. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Trunking
It should work as is, just make usre that you have an extension defined (or a catch all) for every DID you have with the provider so that incoming works. On 6/2/06, Steven Haldeman <[EMAIL PROTECTED]> wrote: Hello, I am attempting to figure out how to set up SIP trunking, between my company and our SIP provider. This is an expermintal project at this time. The SIP provider gave us a Signalling IP address and two Media IP addresses. We supplied them with the IP address of our Asterisk box. When asked what our Usernames and Passwords would be we were told that they were not needed for a SIP trunk. We can use what they call SIP lines that use username/password however because of tarrifing the lines cost more per month than a trunk I have been successfull in making a SIP Line work, but have no idea where to start with a SIP Trunk. We sill be using DID numbers on the SIP trunks. Has anyone had any experiance with this type of configuration an example would be extrememly helpfull. I have search the internet for help and I may have seen a solution but just was not certain what I was looking at, or how to implement as everything that I have seen user a Username/Password combo. Thank you in advance. Thanks Steven __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 ___ --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
Re: [Asterisk-Users] SIP Trunking
Thank you for your response. All that I get when I dial in is all circutes are busy and when I dial out 503 errors. Here are my configs. Any ideas would be greatly appreciated. The provider is using a Tekelec 9000 Class 5 switch if that is any help. The provider sat us up two accounts one that they are calling a line that uses authentication usernames and passwords as though we were terminating into a SIP phone. This account works but is not the prefered solution to our problem. The other account is what the provider calls a trunk account and it does not use usernames and passwords. This seems to be the solution for the provider and us. Thank you, Steven sip.conf
[inbound-trunk]type=friendcontext=incominginsecure=veryhost=xxx.xxx.xxx.xxxoutboundproxy=xxx.xxx.xxx.xxxfromdomain=xxx.xxx.xxx.xxxdefaultip=xxx.xxx.xxx.xxxdisallow=allallow=ulawnat=yescanreinvite=yesqualify=yes extension.conf [incoming]exten => NXXNXX,1,Answer()exten => NXXNXX,2,Background(greeting)exten => NXXNXX,3,SayDigits(${CALLERIDNUM})exten => NXXNXX,4,Dial(SIP/steven)exten => NXXNXX,5,Hangup()C F <[EMAIL PROTECTED]> wrote: It should work as is, just make usre that you have an extensiondefined (or a catch all) for every DID you have with the provider sothat incoming works.On 6/2/06, Steven Haldeman <[EMAIL PROTECTED]>wrote:>>
Hello,>> I am attempting to figure out how to set up SIP trunking, between my company> and our SIP provider. This is an expermintal project at this time. The SIP> provider gave us a Signalling IP address and two Media IP addresses. We> supplied them with the IP address of our Asterisk box. When asked what our> Usernames and Passwords would be we were told that they were not needed for> a SIP trunk. We can use what they call SIP lines that use username/password> however because of tarrifing the lines cost more per month than a trunk I> have been successfull in making a SIP Line work, but have no idea where to> start with a SIP Trunk. We sill be using DID numbers on the SIP trunks.> Has anyone had any experiance with this type of configuration an example> would be extrememly helpfull.>> I have search the internet for help and I may have seen a solution but just> was not
certain what I was looking at, or how to implement as everything> that I have seen user a Username/Password combo.>> Thank you in advance.>> Thanks>> Steven>>> __> Do You Yahoo!?> Tired of spam? Yahoo! Mail has the best spam protection around> http://mail.yahoo.com> ___> --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>>>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options
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Re: [Asterisk-Users] SIP Trunking
I also forgot to mention that our provider gave use three different IP addresses. One IP address for "Signalling" and two addresses for "Media." If this helps any. Thank you, StevenSteven Haldeman <[EMAIL PROTECTED]> wrote:Thank you for your response. All that I get when I dial in is all circutes are busy and when I dial out 503 errors. Here are my configs. Any ideas would be greatly appreciated. The provider is using a Tekelec 9000 Class 5 switch if that is any help. The provider sat us up two accounts one that they are calling a line that uses authentication usernames and passwords as though we were terminating into a SIP phone. This account works but is not the prefered solution to
our problem. The other account is what the provider calls a trunk account and it does not use usernames and passwords. This seems to be the solution for the provider and us. Thank you, Steven sip.conf [inbound-trunk]type=friendcontext=incominginsecure=veryhost=xxx.xxx.xxx.xxxoutboundproxy=xxx.xxx.xxx.xxxfromdomain=xxx.xxx.xxx.xxxdefaultip=xxx.xxx.xxx.xxxdisallow=allallow=ulawnat=yescanreinvite=yesqualify=yes extension.conf [incoming]exten => NXXNXX,1,Answer()exten => NXXNXX,2,Background(greeting)exten => NXXNXX,3,SayDigits(${CALLERIDNUM})exten => NXXNXX,4,Dial(SIP/steven)exten => NXXNXX,5,Hangup()C F <[EMAIL PROTECTED]> wrote:
It should work as is, just make usre that you have an extensiondefined (or a catch all) for every DID you have with the provider sothat incoming works.On 6/2/06, Steven Haldeman <[EMAIL PROTECTED]>wrote:>> Hello,>> I am attempting to figure out how to set up SIP trunking, between my company> and our SIP provider. This is an expermintal project at this time. The SIP> provider gave us a Signalling IP address and two Media IP addresses. We> supplied them with the IP address of our Asterisk box. When asked what our> Usernames and Passwords would be we were told that they were not needed for> a SIP trunk. We can use what they call SIP lines that use username/password> however because of tarrifing the lines cost more per month than a trunk I> have been successfull in making a SIP Line
work, but have no idea where to> start with a SIP Trunk. We sill be using DID numbers on the SIP trunks.> Has anyone had any experiance with this type of configuration an example> would be extrememly helpfull.>> I have search the internet for help and I may have seen a solution but just> was not certain what I was looking at, or how to implement as everything> that I have seen user a Username/Password combo.>> Thank you in advance.>> Thanks>> Steven>>> __> Do You Yahoo!?> Tired of spam? Yahoo! Mail has the best spam protection around> http://mail.yahoo.com> ___> --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>>>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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RE: [Asterisk-Users] sip trunking works?
'SIP trunking' is something I've heard before mainly from traditional switch vendors that are having trouble with SIP as they are used to 'stations' and 'trunks'. You should find asterisk very capable as a 'toolkit' box, as it can 'register' like a traditional SIP client with a SIP Registrar, or act as one, or both, or neither! Its pretty flexible. I'd be interested how you go with the Alcatel. On the SER (www.iptel.org) site they have a couple of page intro on SIP which is worth the read and will put you miles ahead of 90% of the ICT world which has problems spelling SIP :) Good luck, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Hajek Sent: Tuesday, 9 November 2004 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] sip trunking works? Hello, I'm about to connect asterisk with Alcatel Enterprise PBX using SIP trunking, I can't find if Asterisk has this capability. Can you please advice? Thank you. -David ___ 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 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
RE: [Asterisk-Users] sip trunking works?
Are you talking about the 187 page SIP tutorial? What "couple of pages" are you referring to? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Childs Sent: Tuesday, November 09, 2004 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] sip trunking works? 'SIP trunking' is something I've heard before mainly from traditional switch vendors that are having trouble with SIP as they are used to 'stations' and 'trunks'. You should find asterisk very capable as a 'toolkit' box, as it can 'register' like a traditional SIP client with a SIP Registrar, or act as one, or both, or neither! Its pretty flexible. I'd be interested how you go with the Alcatel. On the SER (www.iptel.org) site they have a couple of page intro on SIP which is worth the read and will put you miles ahead of 90% of the ICT world which has problems spelling SIP :) Good luck, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Hajek Sent: Tuesday, 9 November 2004 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] sip trunking works? Hello, I'm about to connect asterisk with Alcatel Enterprise PBX using SIP trunking, I can't find if Asterisk has this capability. Can you please advice? Thank you. -David ___ 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 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 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
RE: [Asterisk-Users] sip trunking works?
Appologies I should have put a link http://www.iptel.org/ser/sipintro.html Its only 20 pages, and they are pretty small pages :) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Rodan Sent: Wednesday, 10 November 2004 5:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sip trunking works? Are you talking about the 187 page SIP tutorial? What "couple of pages" are you referring to? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Childs Sent: Tuesday, November 09, 2004 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] sip trunking works? 'SIP trunking' is something I've heard before mainly from traditional switch vendors that are having trouble with SIP as they are used to 'stations' and 'trunks'. You should find asterisk very capable as a 'toolkit' box, as it can 'register' like a traditional SIP client with a SIP Registrar, or act as one, or both, or neither! Its pretty flexible. I'd be interested how you go with the Alcatel. On the SER (www.iptel.org) site they have a couple of page intro on SIP which is worth the read and will put you miles ahead of 90% of the ICT world which has problems spelling SIP :) Good luck, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Hajek Sent: Tuesday, 9 November 2004 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] sip trunking works? Hello, I'm about to connect asterisk with Alcatel Enterprise PBX using SIP trunking, I can't find if Asterisk has this capability. Can you please advice? Thank you. -David ___ 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 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 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 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
Re: [asterisk-users] SIP Trunking Mantra (Origination)
Interesting. You might want to consider paying some expert for consulting ? Mitul On Jun 22, 2013 7:21 PM, "Nick Khamis" wrote: > Hello Everyone, > > We are currently having talks with various service providers, and > trying to determine what the best way is to interconnect in order to > have access to the PSTN network. As you know there are two ways of > doing this: > > Traditional PRI: Have trunks grouped into a transport layer such as > OC3/12. With DIDs attached to the group. As you many know, this > approach would also require a POP near the CO of the exchange we want > to service etc.. We could also have the service provider backhaul some > of the NXX in areas we do not have a POP, to a location near by. > > SIP Trunking: SIP traffic coming through the end of transport layer > such as OC3 or ethernet connection directly connected to the service > provider, with DID that can come from anywhere. No need for a POP in > Chicago, for example, when we are located in Kansas. > > The benefits of one over the other are known, and not the topic of > this message. What we are trying to determine are: > > When talking market price, a "virtual PRI/SIP Trunk" interconnect > costs about 500-550 per 24 channel virtual pri. This compared to a > true "ISDN/PRI" which can costs between "200-500" dollars depending > who you talk to. We also have to take into consideration the hardware > needed for either setup i.e.: > > * Option 1: SIP Proxy > * Option 2: media gatweays, multiplexers, media server > > Even though it was natural to talk about "pricing", this is still not > what we are interested in knowing. What we are interested in finding > out is: > > * How are service providers that offer "virtual pris" interconnected > with their suppliers? I would imagine that some (non-CLECS), are > renting a connection from the LECs, and grouping PRI/ISDN trunks > (option 2). And others (CLECS), have a A-Link/ISUP trunk interconnect > to the CO. > - Which brings up a second question. How does a PRI trunk group > differ from an ISUP > trunk. I don't know much about and ISUP trunks and would *really* > appreciate having >someone educate us on (i) the concept, (ii) what type of equipment > would be needed, >(iii) how it differs from ISDN trunk groups. (iv) is it only > available for LECS > > I do have more questions, however for the sake of brevity will stop > right here. And before anyone asks the "it depends what you want to > do", I will mention that we are trying to establish an interconnection > that will sustain 2016 channels or 84 T1s, and 5000 DIDs. We are not > trying to become a CLEC however, still feel that option 2 would be the > better choice for reasons covered here, and some that are left > implicit (i.e, quality, reliability of managing our own > networks..). > > Your insights are greatly appreciated! > > Nick. > > -- > _ > -- 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] SIP Trunking Mantra (Origination)
Thank you mitul. N. -- _ -- 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] SIP Trunking Mantra (Origination)
Any other experts out there? -- _ -- 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] SIP Trunking Mantra (Origination)
On 6/25/13, Jai Rangi wrote: > Not a problem, I wanted to tell you the diff between PRI and sip trunking. > I am sure there are lots of option we are just fine what ever works best > for you. > > Back to subject we strongly believe that sip trunking is far better option > than PRI and that's the way to go in future. > > Jai Hello Jai the benefits of SIP trunking is well noted. However, there will always be an underline interconnect that makes SIP trunking possible. What I am trying to say is that there will always be ISUP/ISDN trunk groups that we throw a TCP/IP stack on top, and offer clients with SIP trunking. We are looking for information on how to accomplish interconnect using ISUP trunks (i.e., SS7 interconnect). Kind Regards, Nick. -- _ -- 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] sip trunking and call transfer
Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 <[EMAIL PROTECTED]> wrote: > Hi to all. > > i-ve got a question: > > what happen when a call between 2 trunks is transferred to another trunk? > > For example, suppose that i have 4 trunk A,B,C,D: > > Caller 1 - Trunk A/B - Caller2 > > Then Caller 2 transfer to Caller 3 behind Trunk B/C > > What happend? > > a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 > > or > > b) Caller 1 - Trunk A/C - Caller3 > > So: > > is it possible to avoid the scenario a) ? > > Thanks to all > -- > /*/ > nik600 > http://www.kumbe.it > -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
Maybe because there is no such thing as a "SIP trunk", at least in the Asterisk world. Most of us call them "peer" or "friend". The term you are looking for is "reinvite". Reinvites allow two devices to send audio directly between the two end points of the call. the SIGNALING stays on the servers, but the audio can be sent directly between the two end points. NAT, transcoding, and the T and t options to dial (as well as other things) will prevent reinvies from happening. nik600 wrote: > Maybe my question is not clear or is too stupid? (sorry) > > Maybe this is already done in SIP trunking? > > Or (worste case) is impossible to do that? > > Thanks > > On Fri, Nov 21, 2008 at 8:53 AM, nik600 <[EMAIL PROTECTED]> wrote: >> Hi to all. >> >> i-ve got a question: >> >> what happen when a call between 2 trunks is transferred to another trunk? >> >> For example, suppose that i have 4 trunk A,B,C,D: >> >> Caller 1 - Trunk A/B - Caller2 >> >> Then Caller 2 transfer to Caller 3 behind Trunk B/C >> >> What happend? >> >> a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 >> >> or >> >> b) Caller 1 - Trunk A/C - Caller3 >> >> So: >> >> is it possible to avoid the scenario a) ? >> >> Thanks to all >> -- >> /*/ >> nik600 >> http://www.kumbe.it >> > > > -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
On Sun, Nov 23, 2008 at 5:54 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]>wrote: > The term you are looking for is "reinvite". Reinvites allow two devices > to send audio directly between the two end points of the call. the > SIGNALING stays on the servers, but the audio can be sent directly > between the two end points. This still leaves the SIP signaling hairpin on Caller 2's system. > nik600 wrote: > >> a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 > >> > >> or > >> > >> b) Caller 1 - Trunk A/C - Caller3 > >> > >> So: > >> > >> is it possible to avoid the scenario a) ? Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's session with Caller 2 and send a new INVITE to Caller 3. So, this is how you do it from a SIP protocol perspective. I'm not sure to what extent Asterisk supports this capability. -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
On Mon, Nov 24, 2008 at 12:14 AM, Raj Jain <[EMAIL PROTECTED]> wrote: > > Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller > 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's > session with Caller 2 and send a new INVITE to Caller 3. So, this is how you > do it from a SIP protocol perspective. I'm not sure to what extent Asterisk > supports this capability. > -- > Raj Jain ok, thanks for your reply! I'll search about Asterisk SIP referer implementation. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
