RE: [Asterisk-Users] Multi Asterisk Server Transfers
Hi! > Call is then connected as follows. > > PSTN -> Provider -> Head Office -> Provider -> Remote > > But after it is transferred, I want the resulting route to be: > > PSTN -> Provider -> Remote > > Otherwise Head office has 2 times the bandwidth running through it for a > call not even going to one of it's own extensions. I had throught that the > IAX connection between Provider and Head Office would "pass off" calls that > way. It should - you need to find out if either the office or the provider * has "notransfer=yes" in the iax.conf file. If yes: remove that. Take a careful look at the IAX log and debug messages to find out when/if Asterisk tries to do a native transfer (and probably fails for some reason). Next to this you need to check that the codecs are matching, your best bet is to use the same codec on all sides. In the worst case you have exactly one of the clients talking g729 and only one * box that has a transcoding license for this. By the way, with all this I/we assume that you are using the Asterisk # transfer mechanism and not a phone-based one. Cheers, Philipp ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
> Anyone know a good IAX phone (not softphone)? Only phones I know that support IAX can be found there : http://www.iaxtalk.com/index.php ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
Philipp von Klitzing wrote: Instead I'd go for a co-located Asterisk that the remote SIP devices register with, and then link both * boxes (co-located and central office) using IAX2 with IAX native transfers enabled. Of course this means that the office * _only_ talks IAX and that all calls to the remote SIP clients _always_ go thru the co-located box (with its extra bandwidth). Erm that's the assumption I was making, there was a centralised box somewhere... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "I do not try to dance better than anyone else. I only try to dance better than myself." ___ 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
RE: [Asterisk-Users] Multi Asterisk Server Transfers
Simple as that? Anyone know a good IAX phone (not softphone)? Thanks Mike >Then you need to use the same protocol to the provider. One office is >using SIP, the other is using IAX. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
Mike Sander wrote: I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office transfers to 0 where is SIP extension according to Provider and 0 is to dial out on the trunk. Call is then connected as follows. PSTN -> Provider -> Head Office -> Provider -> Remote But after it is transferred, I want the resulting route to be: PSTN -> Provider -> Remote Otherwise Head office has 2 times the bandwidth running through it for a call not even going to one of it's own extensions. I had throught that the IAX connection between Provider and Head Office would "pass off" calls that way. Let me know, but thanks for all the help so far. Then you need to use the same protocol to the provider. One office is using SIP, the other is using IAX. ___ 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
RE: [Asterisk-Users] Multi Asterisk Server Transfers
I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office transfers to 0 where is SIP extension according to Provider and 0 is to dial out on the trunk. Call is then connected as follows. PSTN -> Provider -> Head Office -> Provider -> Remote But after it is transferred, I want the resulting route to be: PSTN -> Provider -> Remote Otherwise Head office has 2 times the bandwidth running through it for a call not even going to one of it's own extensions. I had throught that the IAX connection between Provider and Head Office would "pass off" calls that way. Let me know, but thanks for all the help so far. Mike >Instead I'd go for a co-located Asterisk that the remote SIP devices >register with, and then link both * boxes (co-located and central office) >using IAX2 with IAX native transfers enabled. Of course this means that >the office * _only_ talks IAX and that all calls to the remote SIP >clients _always_ go thru the co-located box (with its extra bandwidth). >SER certainly is another way to go (as mentioned before), but in this >specific setup I assume it complicates matters unnecessarily. >Cheers, Philipp -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
Hi! > > Is this possible? Companies with multiple * servers in many remote office, > > surely have this system, to conserve bandwidth? How is the transfer made? > > Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] > > basic > > release. > > Simple way is to use the # transfers in asterisk on the main box then > don't allow transfers on the remote boxes and don't use transfer buttons... There is an even simpler solution: Don't make phone calls. ;-> Seriously, I think the poster _requires_ transfers in the first place, so this your reply won't help him too much. Instead I'd go for a co-located Asterisk that the remote SIP devices register with, and then link both * boxes (co-located and central office) using IAX2 with IAX native transfers enabled. Of course this means that the office * _only_ talks IAX and that all calls to the remote SIP clients _always_ go thru the co-located box (with its extra bandwidth). SER certainly is another way to go (as mentioned before), but in this specific setup I assume it complicates matters unnecessarily. Cheers, Philipp ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
Mike Sander wrote: Is this possible? Companies with multiple * servers in many remote office, surely have this system, to conserve bandwidth? How is the transfer made? Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] basic release. Simple way is to use the # transfers in asterisk on the main box then don't allow transfers on the remote boxes and don't use transfer buttons... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "I do not try to dance better than anyone else. I only try to dance better than myself." ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
On Wed, 26 Jan 2005 22:42:52 -0800, Luki wrote: >> if it were my project, I'd look into Asterisk on a small form >> factor/embedded system like a Soekris Engineering box > >In that case I guess you could use a Linksys WRT54G instead and run * >on it :-). Comes fully assembled (read: with a case) and a 4-port >switch to connect the phone and a computer... and can probably even do >QOS to prioritize voice traffic. Never mind the WiFi part, but might >be handy as well. > >Someone asked WHY you would want to run * on a simple WRT54G, I guess >this is a possible scenario. Perhaps. I personally don't like the idea of using the WRT54G. I've been burned by Linksys in the past. I run m0n0wall on Soekris 4501. The hardware with case cost me $<200. I understand that WRAP boards are less than Soekris. You could, as an alternative, buy one of the new routers with SIP ATA capability and have it log into the ITSP and head office *. I'm not sure if they support multiple registries. But again, I'd prefer a real SIP phone with business class features. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
> if it were my project, I'd look into Asterisk on a small form > factor/embedded system like a Soekris Engineering box In that case I guess you could use a Linksys WRT54G instead and run * on it :-). Comes fully assembled (read: with a case) and a 4-port switch to connect the phone and a computer... and can probably even do QOS to prioritize voice traffic. Never mind the WiFi part, but might be handy as well. Someone asked WHY you would want to run * on a simple WRT54G, I guess this is a possible scenario. --Luki ___ 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
RE: [Asterisk-Users] Multi Asterisk Server Transfers
On Thu, 27 Jan 2005 14:24:30 +1100, Mike Sander wrote: >I agree with you. If every office had a * server, it would be fine. > >i.e. Office 1 rings office 2, then gets transferred to office 3, then >connection is direct from office 1 to 3, and 2 releases all contact. > >However, what if office 3 is a 1 person office, with just a single SIP phone >connected to the VoIP provider. Full IAX trunking can do hand-offs quite >simply, I think, but when the destination is a single SIP connection, things >get messy. > >Is this relevant to your answer, because I'm a little confused now? > >With thanks > If one office has only a single SIP phone then you don't trunk to them via IAX. You let them connect directly to the ITSP for their outgoing/incomming PSTN calls. You program dialplan logic on your * boxen to connect to that SIP extension via IP. You could get them a free SIP account like FWD, SIPPhone, etc and use that to setup presets for calling from that extension to your other offices directly via IP. It'd take more of a phone than a Grandstream, but then they're pretty simple for business use anyway. I like my Polycom IP600. I also like my Zultys 4x5. If the end, if it were my project, I'd look into Asterisk on a small form factor/embedded system like a Soekris Engineering box...even for single extension offices. That'd give you a low cost way to offer comparable programmability and dialplan logic in all cases. Also local voicemail, if that's of any value. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
RE: [Asterisk-Users] Multi Asterisk Server Transfers
I agree with you. If every office had a * server, it would be fine. i.e. Office 1 rings office 2, then gets transferred to office 3, then connection is direct from office 1 to 3, and 2 releases all contact. However, what if office 3 is a 1 person office, with just a single SIP phone connected to the VoIP provider. Full IAX trunking can do hand-offs quite simply, I think, but when the destination is a single SIP connection, things get messy. Is this relevant to your answer, because I'm a little confused now? With thanks Mike >Seems strange to be handling multiple * servers over SIP and ignoring >IAX2. I'd be inclined to trunk between offices over IAX2. In fact, I'd >use and IAX2 based ITSP and then be able to hand off calls in a >.reinvite fashion without all the messy port handling. >In addition you save on bandwidth by trunking multiple calls over one >IAX2 connection. Less IP overhead, between offices and to the ITSP. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
On Thu, 27 Jan 2005 12:58:56 +1100, Mike Sander wrote: >Hi, > >We are in the business of setting up * servers for businesses, attached via >IAX trunks to our VoIP provider (also using *). > >I have a client with a head office * server, who wants a number of remote >offices, with just 1 SIP connection to each. I can arrange this no probs >with our providers, but there are issues with transfer. > >I don't want the remote offices making their direct SIP connection to the >head office, because bandwidth is limited and then for them to make an >outgoing call, the head office has both an incoming and outgoing connection >- or double the bandwidth. This is the same for an incoming call to head >office that gets transferred to the remote, the call stays with the head >office * server, and the server makes another outgoing call to the remote >office. All these calls are free, but use double the bandwidth. > >The question: > >The remote offices can make direct SIP connections to our provider. If the >head office * transfers a call, then the server releases the call entirely >back to the providers * server and calls from there. > >I.E. call in to head office from PSTN through the provider. Call gets >transferred to the remote office. Head office could then unplug/burn/blowup >their asterisk server without disrupting the call between the remote office >and the PSTN network. > >Is this possible? Companies with multiple * servers in many remote office, >surely have this system, to conserve bandwidth? How is the transfer made? >Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] basic >release. > Seems strange to be handling multiple * servers over SIP and ignoring IAX2. I'd be inclined to trunk between offices over IAX2. In fact, I'd use and IAX2 based ITSP and then be able to hand off calls in a reinvite fashion without all the messy port handling. In addition you save on bandwidth by trunking multiple calls over one IAX2 connection. Less IP overhead, between offices and to the ITSP. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
Re: [Asterisk-Users] Multi Asterisk Server Transfers
Sounds like you need a SIP Proxy, which just relays calls to the destination, rather than Asterisk which handles all aspects of a call. I'd recommend Sip Express Router, http://www.iptel.org/ On Thu, 27 Jan 2005 12:58:56 +1100, Mike Sander <[EMAIL PROTECTED]> wrote: > Hi, > > We are in the business of setting up * servers for businesses, attached via > IAX trunks to our VoIP provider (also using *). > > I have a client with a head office * server, who wants a number of remote > offices, with just 1 SIP connection to each. I can arrange this no probs > with our providers, but there are issues with transfer. > > I don't want the remote offices making their direct SIP connection to the > head office, because bandwidth is limited and then for them to make an > outgoing call, the head office has both an incoming and outgoing connection > - or double the bandwidth. This is the same for an incoming call to head > office that gets transferred to the remote, the call stays with the head > office * server, and the server makes another outgoing call to the remote > office. All these calls are free, but use double the bandwidth. > > The question: > > The remote offices can make direct SIP connections to our provider. If the > head office * transfers a call, then the server releases the call entirely > back to the providers * server and calls from there. > > I.E. call in to head office from PSTN through the provider. Call gets > transferred to the remote office. Head office could then unplug/burn/blowup > their asterisk server without disrupting the call between the remote office > and the PSTN network. > > Is this possible? Companies with multiple * servers in many remote office, > surely have this system, to conserve bandwidth? How is the transfer made? > Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] > basic > release. > > Thanks > Mike > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 > > ___ > 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 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] Multi Asterisk Server Transfers
Hi, We are in the business of setting up * servers for businesses, attached via IAX trunks to our VoIP provider (also using *). I have a client with a head office * server, who wants a number of remote offices, with just 1 SIP connection to each. I can arrange this no probs with our providers, but there are issues with transfer. I don't want the remote offices making their direct SIP connection to the head office, because bandwidth is limited and then for them to make an outgoing call, the head office has both an incoming and outgoing connection - or double the bandwidth. This is the same for an incoming call to head office that gets transferred to the remote, the call stays with the head office * server, and the server makes another outgoing call to the remote office. All these calls are free, but use double the bandwidth. The question: The remote offices can make direct SIP connections to our provider. If the head office * transfers a call, then the server releases the call entirely back to the providers * server and calls from there. I.E. call in to head office from PSTN through the provider. Call gets transferred to the remote office. Head office could then unplug/burn/blowup their asterisk server without disrupting the call between the remote office and the PSTN network. Is this possible? Companies with multiple * servers in many remote office, surely have this system, to conserve bandwidth? How is the transfer made? Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] basic release. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 ___ 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
