[asterisk-users] IAX or SIP - connecting two Asterisk servers together
Hello Folks; Again, excuse my cluelessness. I have an Asterisk server in the US - and I want to connect it to one in Europe. Here is my scenario: 1. call a phone number, my Asterisk box in the US answers 2. perhaps a 'please wait' voice message 3. it dials an extension on the other Asterisk box in Europe. I am not looking for someone to do this for me, I am just not really sure how to get started. Perhaps some suggested reading, examples, etc? Any help at all would be appreciated. Thanks much. Glen -- _ -- 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] IAX or SIP - connecting two Asterisk servers together
On Tue, Nov 02, 2010 at 03:20:48PM -0400, Silver Thorne wrote: I am not looking for someone to do this for me, I am just not really sure how to get started. Perhaps some suggested reading, examples, etc? The simplest approach would be to skip the answering and just dial through immediately, feeding back the destination's ring tone to the originator. Set up an IAX link between the two boxes (you could do it with SIP, but I found IAX less trouble), then set up an appropriate bit of dialplan logic on the American box, as it might be: exten = 4682,1,Dial(IAX2/usern...@eurobox/8873) Roger -- _ -- 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] IAX or SIP - connecting two Asterisk servers together
Silver Thorne wrote: Hello Folks; Again, excuse my cluelessness. I have an Asterisk server in the US - and I want to connect it to one in Europe. Here is my scenario: 1. call a phone number, my Asterisk box in the US answers 2. perhaps a 'please wait' voice message 3. it dials an extension on the other Asterisk box in Europe. I am not looking for someone to do this for me, I am just not really sure how to get started. Perhaps some suggested reading, examples, etc? Any help at all would be appreciated. Thanks much. Glen This is fairly easy to do, and works well using the IAX protocol. There are a group of some 100-200 nodes worldwide doing something similar connecting electromechanical switches and collectors ( think amateur radio without the radio ) The way we are doing it wouldn't help you too much as we have a central reference point for line numbers, but there are nodes from NZ to the UK to the US that are interconnected on an as needed basis. Our dialplan is based on ( in the US ) the NANP with no NPA, and outside the US existing country codes. We have had some amusing examples of hackers that have managed to break in and thinking they have PSTN access, ended up with some ( to them ) surprises. Call setup times are short, 1-2 seconds generally, so I doubt you would even need a please wait recording. See ckts.info for more information on the network and some coding examples that might help. John Novack -- 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] IAX or SIP - connecting two Asterisk servers together
On Tue, Nov 2, 2010 at 3:20 PM, Silver Thorne zora...@gmail.com wrote: Any help at all would be appreciated. DUNDi -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] IAX vs SIP
Hi, Just to review the test I did: ---SIP extension-- Trunk - | SIPp |---| Asterisk 1 || Asterisk 2 | ------ -- Both Asterisk boxes are virtual machines in VirtualBox and version 1.4.21.2. I generated calls using SIPp, and I monitored the cpu utilization in the Asterisk 1 with top. I compare the cpu utilization when I used IAX and when I used SIP as Trunk protocol. The following are the results (averages) I got with ulaw codecs in both sides: Calls IAX SIP 4 6,0 1,8 10 9,2 4,6 20 19,58,6 30 28,213,5 34 36,916,2 40 38,219,5 50 36,924,3 As you can see, IAX almost doubles the cpu cycles. I repeated the test using gsm as the trunk codec, and in this scenario IAX shows a better performance (Sip extension continues with ulaw): IAX SIP 1,8 25,7 12,841,5 29,247,0 45,769,4 54,278,5 53,383,3 65,787,1 And finally I repeated with iLBC in the trunk, and SIP won again: IAX SIP 8,5 9,4 29,314,5 57,623,6 74,437,3 84,341,5 -- 51,2 -- 67,0 Does this makes sense? Any feedback? Has anybody done similar test for comparison? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP
On 7 Sep 2008, at 21:34, Edgar Guadamuz wrote: Hello, I have been testing a trunk IAX and another SIP, using sipp to generate SIP calls to a Asterisk box. The testing dialplan just connects to another Asterisk box, who answers the call and playback some files. I noticed that the cpu load is higher when I use an IAX, about 90% for 25 simultaneous calls. In the other hand, with a SIP trunk the cpu load was about the half or less. In both cases the Asterisk box was in the middle of the RTP path, and both the trunk and the sip client had the same codec, ulaw. Does it make sense? Why is IAX demanding so much cpu load? Which Asterisk version are you running? There was a specific version (1.4.20 I think) that had made IAX super-expensive. The most recent versions of asterisk _should_ have IAX being roughly equivalent in CPU usage as SIP Incidentally if anyone has comparative numbers for IAX vs SIP on 1.6 betas (or hyper-recent 1.4) I'd love to have them for a talk I'm doing at astricon. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX vs SIP
Hello, I have been testing a trunk IAX and another SIP, using sipp to generate SIP calls to a Asterisk box. The testing dialplan just connects to another Asterisk box, who answers the call and playback some files. I noticed that the cpu load is higher when I use an IAX, about 90% for 25 simultaneous calls. In the other hand, with a SIP trunk the cpu load was about the half or less. In both cases the Asterisk box was in the middle of the RTP path, and both the trunk and the sip client had the same codec, ulaw. Does it make sense? Why is IAX demanding so much cpu load? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX vs SIP - Getting Asterisk out of the media path
If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at both ends) but the media continues passing through my network. The default behaviour for the Dial command is to have Asterisk step out of the media path provided you avoid some options like tT, which I do, so this should work. One interesting note: In an Ethereal trace, I see 407 Proxy Authentication required just after the INVITE to the callee. Could that be part of the problem? If so what's the fix? I thought it had something to do with the auth parameter. I am: - Behind a NAT, - Running Red Hat 9.0 - Running Asterisk 1.2.14 How do I stop the media passsing through my Asterisk server after a call between two external parties has been bridged? My sip.conf and the dial command I use are below. Thanks, Hugh ;*** Dial Command *** exten = _6136930630,n,Dial(SIP/[EMAIL PROTECTED]) ; SIP.conf ** [general] ; context=incoming-bogus-calls bindport=5060 bindaddr=0.0.0.0 maxexpirey=3600 defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; externip=999.99.999.99 ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; register=6135551234:[EMAIL PROTECTED]/6135551234 ; [6135551234] type=peer ;auth=md5 auth=6135551234:[EMAIL PROTECTED] username=6135551234 fromuser=6135551234 fromdomain=myITSP.ca secret= host=sip02.myITSP.ca port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=very context=incoming-sip -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.36/681 - Release Date: 11/02/2007 ___ --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] IAX vs SIP trunks between Asterisk boxes
Brad Templeton wrote: On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote: Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) The bandwidth of the audio stream dwarfs the bandwidth of signalling traffic by orders of mangitude. So in fact, I think this is exactly wrong. If bandwidth to or between the servers is a concern, that's where you most want to not be in the audio path. But if you have multiple RTP streams emnbedded in an IAX trunk, then the IP overhead is significantly reduced. AFAIK video should work for IAX2, there is explicit support for it. (unlike h323). Asterisk will only be able to pass the raw RTP traffic though, since it doesn't have any video codecs (just format definitions). ___ --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] IAX vs SIP trunks between Asterisk boxes
Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) It is worth remembering in this sort of setup, often the phones at one site will not have a route to the phons on the other site, so the calls wont be re-invited off to the handsets anyway. ___ --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] IAX vs SIP trunks between Asterisk boxes
Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) As I understand it video will NOT work if you use an IAX trunk between * boxes, it must be SIP. Just food for thought in case you are planning on using video. David ___ --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] IAX vs SIP trunks between Asterisk boxes
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote: Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) The bandwidth of the audio stream dwarfs the bandwidth of signalling traffic by orders of mangitude. So in fact, I think this is exactly wrong. If bandwidth to or between the servers is a concern, that's where you most want to not be in the audio path. It is worth remembering in this sort of setup, often the phones at one site will not have a route to the phons on the other site, so the calls wont be re-invited off to the handsets anyway. If it's phone-on-NAT to phone-on-different-NAT, it typically will not work. That doesn't mean it can't work if bandwidth is important. I think the complete solution, not yet in Asterisk as I understand it is for Asterisk to be aware of both the internal and external addresses of a phone, and to connect internal phones with their internal addresses, but to connect internal phones to external endpoints through their external addresses. Ideally audio never flows through asterisk unless it's doing an IVR dialogue or otherwise explicitly wants it to. (In fact, ideally DTMF goes via SIP INFO or its successors so that Asterisk can listen to the DTMF without being in on the audio.) Flowing audio through your box costs not just bandwidth, it adds latency, and very slight extra risks of packet loss. Latency is the bane of voip calls, it also worsens echo. ___ --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] IAX vs SIP trunks between Asterisk boxes
On Fri, Jan 05, 2007 at 11:33:02AM +, Gordon Henderson wrote:
On Thu, 4 Jan 2007, Noah Miller wrote:
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then
sent
to the UA from the second box with SIP?
DTMF passthrough issues?
I've got a client with sip phones on several different servers and
IAX links between the servers, so I guess that's pretty similar to
your setup. I've never bothered to check for overhead since it was
never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram,
with never more than 3-4 calls going through any one of the IAX
links). I can say that DTMF works fine in this setup.
I'm doing the same on 1GHz processors - CPU usage is virtually nil unless
there's transcoding going on (about 4% per GSM transcode)
ADSL bandwidth is more of a concern for me in these applications )-:
While it would be work to set up, you actually ideally want to
trunk with the same protocol being used by the external phones
or endpoints. When connecting a SIP to SIP call (presuming you
don't have annoying nat problems or have turned canreinvite off)
the audio should go directly from endpoint to endpoint and not
via asterisk.Ditto on IAX to IAX calls.
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
In some ways, an ideal solution would have two trunk connections
between the boxes (really just two config entries in iax.conf and
sip.conf) and go between the boxes with whatever protocol the
calling channel is using. You could write dialplan scripts to
pull out the channel and choose the right * to * protocol (as
opposed to inter-asterisk protocol which has another meaning.
:-)
It can also be worth having a termination provider that you
can talk to with both IAX and SIP, and sending them the call
with the same protocol the phone used.
Annoyingly, IAX and SIP channels use different interfaces
to provide the address, so you can't do
DIAL(${chantype}/[EMAIL PROTECTED])
A cute patch would be to support that with a consistent syntax over
channels.
Note if you use various flags on Dial which require asterisk
to hear dtmf or do other audio, you are stuck hairpinning.
___
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Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
On Thu, 4 Jan 2007, Noah Miller wrote: Hi Damon - Can anyone comment on the overhead added when a SIP call comes into one asterisk box, is routed to another with IAX instead of SIP, and is then sent to the UA from the second box with SIP? DTMF passthrough issues? I've got a client with sip phones on several different servers and IAX links between the servers, so I guess that's pretty similar to your setup. I've never bothered to check for overhead since it was never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram, with never more than 3-4 calls going through any one of the IAX links). I can say that DTMF works fine in this setup. I'm doing the same on 1GHz processors - CPU usage is virtually nil unless there's transcoding going on (about 4% per GSM transcode) ADSL bandwidth is more of a concern for me in these applications )-: Gordon ___ --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 vs SIP trunks between Asterisk boxes
In order to work around some authentication issues I am considering connecting two asterisk boxes with IAX instead of SIP. The original reason for choosing SIP was to reduce the need to translate SIP signaling to IAX, since all origination, termination, and UAs are SIP. Can anyone comment on the overhead added when a SIP call comes into one asterisk box, is routed to another with IAX instead of SIP, and is then sent to the UA from the second box with SIP? DTMF passthrough issues? Any other issues? ___ --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] IAX vs SIP trunks between Asterisk boxes
Hi Damon - Can anyone comment on the overhead added when a SIP call comes into one asterisk box, is routed to another with IAX instead of SIP, and is then sent to the UA from the second box with SIP? DTMF passthrough issues? I've got a client with sip phones on several different servers and IAX links between the servers, so I guess that's pretty similar to your setup. I've never bothered to check for overhead since it was never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram, with never more than 3-4 calls going through any one of the IAX links). I can say that DTMF works fine in this setup. - Noah ___ --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] IAX or SIP termination provider that reaches6421xxxxxxx?
Thank you for your tests! Moj John Marvin wrote: The problem is that most people aren't going to be able to answer this question without trying it. Most voip providers (including Teliax) advertise a rate to all New Zealand Mobile service providers, i.e. +64 2, not specifically +64 21xxx. Note, I just tried a +6421 mobile number via Teliax from the U.S. and it worked. So either 1) Teliax can't reliably connect to those numbers, 2) They can connect to some subset of those numbers, or 3) they fixed something since you last checked. John ___ --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 !DSPAM:500,4513817e124061822916521! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] IAX or SIP termination provider that reaches6421xxxxxxx?
Confirmed, I removed the r from my dial command and it connects now. Thank you for the tip! Luki wrote: I'm interested if anyone else in the Asterisk list can get through to +1-907-747-8633 via voip Sure, no problem. A nice friendly female voice tells you the time and temp, indeed. The thing is that the call never connects -- that info is sent via call progress, so a misconfigured server (i.e. one that uses the r option in dial() or equivalent) would just give you ringing and ringing... [Sep 21 17:49:45] -- Called [EMAIL PROTECTED] [Sep 21 17:49:45] -- SIP/trunks-094da090 is making progress passing it to SIP/1001-b7a030f8 [Sep 21 17:49:48] -- Ringing [Sep 21 17:49:48] -- Progress [Sep 21 17:49:48] -- Peer audio RTP is at port 1.2.3.4:12345 etc. --Luki ___ --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 !DSPAM:500,4513345e104376192314210! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] IAX or SIP termination provider that reaches6421xxxxxxx?
The problem is that most people aren't going to be able to answer this question without trying it. Most voip providers (including Teliax) advertise a rate to all New Zealand Mobile service providers, i.e. +64 2, not specifically +64 21xxx. Note, I just tried a +6421 mobile number via Teliax from the U.S. and it worked. So either 1) Teliax can't reliably connect to those numbers, 2) They can connect to some subset of those numbers, or 3) they fixed something since you last checked. John ___ --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] IAX or SIP termination provider that reaches 6421xxxxxxx?
Is there anybody else out there that can terminate to 6421xxx? It's mobile, New Zealand Vodafone. I'd like to consider all options. Thanks Mojo with Horan Company, LLC wrote: Hi, my asterisk is set up with a pay-as-you-go Teliax account, and can dial out just fine to most numbers, but this cell phone number in New Zealand, 6421xxx, just rings and rings. Teliax support says: Unfortunately, not all International Cell numbers can be dialed by Teliax users. There is a problem from the receiving side. They have restricted us. We may appear as a solicitor to them and that is the way they take the call. If this works from a land line or pots line, that may be the case. This does work from a pots line. Do any list members know of a SIP or IAX termination providers that can call this country/city code combination? the city code, assigned to Vodafone (mobile/wireless?) is 21. Thanks! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] IAX or SIP termination provider that reaches6421xxxxxxx?
Hi Mojo Got through to Chris in NZ using Faktortel no problems. At least if you cant find a NZ voip service who can get through you know an Australian service that can. But isn't the other number a USA number? If so you are totally correct, neither of my US based services were able to connect ( both packet 8 and manhattan virtuals iax service just ring and ring with no pickup) Super interesting because I just dialled with my cingular cell phone after writing this and works like a charm. I'm interested if anyone else in the Asterisk list can get through to +1-907-747-8633 via voip Cheers, Dean (ps I edited the NZ number so don't bother trying) -Original Message- From: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED] Sent: Thursday, 21 September 2006 4:00 PM To: Dean Collins Subject: Re: [asterisk-users] IAX or SIP termination provider that reaches6421xxx? Dean, two numbers that teliax just can't seem to be able to terminate are: 1 9077478633 (time and temperature, not a human) 011 64210594### (vodafone mobile cellphone in new zealand, boss's wife, her name is Chris -- you can tell her you are offering tech support testing the connection for Mojo) Thank you for your help :) Moj Dean Collins wrote: Mojo, whats the number and I'll call it in a few hours (eg during business hours) I use www.Faktortel.com.au via IAX in Australia I've never had a number that I couldn't call in Australia from that service (personally I think the Teliax answer is bullshit) Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, 21 September 2006 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX or SIP termination provider that reaches6421xxx? Is there anybody else out there that can terminate to 6421xxx? It's mobile, New Zealand Vodafone. I'd like to consider all options. Thanks Mojo with Horan Company, LLC wrote: Hi, my asterisk is set up with a pay-as-you-go Teliax account, and can dial out just fine to most numbers, but this cell phone number in New Zealand, 6421xxx, just rings and rings. Teliax support says: Unfortunately, not all International Cell numbers can be dialed by Teliax users. There is a problem from the receiving side. They have restricted us. We may appear as a solicitor to them and that is the way they take the call. If this works from a land line or pots line, that may be the case. This does work from a pots line. Do any list members know of a SIP or IAX termination providers that can call this country/city code combination? the city code, assigned to Vodafone (mobile/wireless?) is 21. Thanks! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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 !DSPAM:500,4512e1be72478243984593! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] IAX or SIP termination provider that reaches6421xxxxxxx?
I'm interested if anyone else in the Asterisk list can get through to +1-907-747-8633 via voip Sure, no problem. A nice friendly female voice tells you the time and temp, indeed. The thing is that the call never connects -- that info is sent via call progress, so a misconfigured server (i.e. one that uses the r option in dial() or equivalent) would just give you ringing and ringing... [Sep 21 17:49:45] -- Called [EMAIL PROTECTED] [Sep 21 17:49:45] -- SIP/trunks-094da090 is making progress passing it to SIP/1001-b7a030f8 [Sep 21 17:49:48] -- Ringing [Sep 21 17:49:48] -- Progress [Sep 21 17:49:48] -- Peer audio RTP is at port 1.2.3.4:12345 etc. --Luki ___ --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 or SIP termination provider that reaches 6421xxxxxxx?
Hi, my asterisk is set up with a pay-as-you-go Teliax account, and can dial out just fine to most numbers, but this cell phone number in New Zealand, 6421xxx, just rings and rings. Teliax support says: Unfortunately, not all International Cell numbers can be dialed by Teliax users. There is a problem from the receiving side. They have restricted us. We may appear as a solicitor to them and that is the way they take the call. If this works from a land line or pots line, that may be the case. This does work from a pots line. Do any list members know of a SIP or IAX termination providers that can call this country/city code combination? the city code, assigned to Vodafone (mobile/wireless?) is 21. Thanks! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] IAX or SIP termination provider that reaches 6421xxxxxxx?
Thank you! looking into it :) kjcsb wrote: We offer NZ mobile termination for NZD0.35/minute (incl GST). If you're based outside New Zealand GST will not be charged so the rate would be NZD0.3111/minute. If you're interested, get a web login at www.conversant.co.nz http://www.conversant.co.nz, then get a ConverseVoice account (no charge). Then contact me and I'll set up the rest. Regards Cameron - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 20 September, 2006 10:28:54 AM Subject: [asterisk-users] IAX or SIP termination provider that reaches 6421xxx? Hi, my asterisk is set up with a pay-as-you-go Teliax account, and can dial out just fine to most numbers, but this cell phone number in New Zealand, 6421xxx, just rings and rings. Teliax support says: Unfortunately, not all International Cell numbers can be dialed by Teliax users. There is a problem from the receiving side. They have restricted us. We may appear as a solicitor to them and that is the way they take the call. If this works from a land line or pots line, that may be the case. This does work from a pots line. Do any list members know of a SIP or IAX termination providers that can call this country/city code combination? the city code, assigned to Vodafone (mobile/wireless?) is 21. Thanks! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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 All new Yahoo! Mail http://us.rd.yahoo.com/mail/uk/taglines/default/nowyoucan/pc_mag/*http://us.rd.yahoo.com/evt=40565/*http://uk.docs.yahoo.com/nowyoucan.html The new Interface is stunning in its simplicity and ease of use. - PC Magazine !DSPAM:500,45107817180481602910205! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] iax vs. sip?
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros and cons. We support both and whilst we have more customers on SIP than IAX, currently favour IAX for new customers where they are undecided given lower support overhead and simplified load-balancing. I'd recommend you try both with the provider you're considering. Simonwww.esms.comOn 8/31/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] iax vs. sip?
We've been using iax with teliax.com for a couple of years, and it seems the quality of calls varies with time. Sometimes it is good and next time its not so good. There has been changes occurring to iax and the jitterbuffer stuff over the last two years, and I'm reasonably certain that some poor quality is related to differences between teliax.com's implementation (eg, s/w versions) and ours. I've not bother to try sip since our asterisk implementation is truly both a production box for our small office, and a test box for various version testing, etc. We used iax for more than a year and moved to sip about 6 months ago. The quality from termination providers seems much better now with sip. Tom At 09:38 PM 8/30/2006, you wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIP might be better to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. ___ --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 vs. sip?
I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --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] iax vs. sip?
We used iax for more than a year and moved to sip about 6 months ago. The quality from termination providers seems much better now with sip. Tom At 09:38 PM 8/30/2006, you wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C6CCA6.8EFA1438 I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIP might be better to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.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
[Asterisk-Users] IAX to SIP conversion: SIP From header issue
I am trying to setup call between IAX client and SIP client using Asterisk 1.2.4. When IAX client makes a call, Asterisk sends SIP invite to the other end point. But, in the From header field, I see -- From: iaxComm User sip:[EMAIL PROTECTED] Is there a way I can have the original username used by IAX client to appear in the from header field of SIP invite? For example, If bob calls Alice using iaxComm, I want sip:[EMAIL PROTECTED] to appear in the SIP invite that is sent to Alice (on her SIP client). In my setup, I use two Asterisks: Bob-Asterisk A--Asterisk BAlice Asterisk A is configured to call Asterisk B on SIP channel when it receives call for Alice. Asterisk B is looking in FROM header field to see who the call is coming from. Since From contains 7, it has no way to find out that it is coming from Bob. Thanks, Jim ___ --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] IAX to SIP conversion: SIP From header issue
JS wrote: I am trying to setup call between IAX client and SIP client using Asterisk 1.2.4. When IAX client makes a call, Asterisk sends SIP invite to the other end point. But, in the From header field, I see -- From: iaxComm User sip:[EMAIL PROTECTED] Is there a way I can have the original username used by IAX client to appear in the from header field of SIP invite? For example, If bob calls Alice using iaxComm, I want sip:[EMAIL PROTECTED] to appear in the SIP invite that is sent to Alice (on her SIP client). In my setup, I use two Asterisks: Bob-Asterisk A--Asterisk BAlice Asterisk A is configured to call Asterisk B on SIP channel when it receives call for Alice. Asterisk B is looking in FROM header field to see who the call is coming from. Since From contains 7, it has no way to find out that it is coming from Bob. Thanks, Jim Jim, Set callerid= for the IAX user/friend. -- Kristian Kielhofner ___ --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] IAX to SIP conversion: SIP From header issue
Jim,Set callerid= for the IAX user/friend.--Kristian Kielhofner This works. Thanks. ___ --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] IAX or SIP
this can be usefull for you. http://www.voip-info.org/wiki-IAX+versus+SIP I guess that for Voice Over IP only, IAX is by far a better choice, since SIP is designed for any kind of session, not just voip calls. IAX is most efficient using bandwidth, you just need 1 port, and works behind firewalls without troubles. But AFAIK does not have videoconference support and that stuff. Best RegardsOn 12/6/05, Ross C [EMAIL PROTECTED] wrote: I recently signed up with TelIAX for voip service. I'm currently using IAX to connect to them. Would connecting to them via SIP provide less latency or be better in any way? Thanks for everyone's 2 cents!! I just to be sure I'm using the best/fastest trunk methods. -Ross ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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 or SIP
I recently signed up with TelIAX for voip service. Im currently using IAX to connect to them. Would connecting to them via SIP provide less latency or be better in any way? Thanks for everyones 2 cents!! I just to be sure Im using the best/fastest trunk methods. -Ross ___ --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 vs SIP (music on hold)
Does IAX support music on hold? It seems only my SIP phones do. Is this correct? ___ 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] IAX vs SIP (music on hold)
I hope I didnt get your question wrong, but if you are asking whether Asterisk can play MOH to an IAX client, then the answer is yes. We have a couple of IAX clients connecting into the queue and are being played MOH while waiting for an operations. Hope this helps :) Cheers On Tue, 29 Mar 2005 14:49:26 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Does IAX support music on hold? It seems only my SIP phones do. Is this correct? ___ 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
Re: [Asterisk-Users] IAX vs SIP (music on hold)
[EMAIL PROTECTED] wrote: Does IAX support music on hold? It seems only my SIP phones do. Is this correct? As I understand it, once the call is delivered to asterisk, it becomes abstracted into a channel. And you can do anything to one channel that you can do to other channels ( with a few notable exceptions including zap channels ). So it shouldn't make a difference whether it's sip/iax/zap as far as MoH is concerned. What may cause issues is what class of MoH is specified, by default and otherwise. But as I haven't tinkered with that a great deal yet, I can't tell you much beyond that. Good luck Sean ___ 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] IAX or SIP answering services
Hello, Does anyone know if it there is an answering service based in the United States that will allow an incoming call through either IAX or sip? Also we would like for them to take orders over the service. Thanks, Linn ___ 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] IAX or SIP
I get a little confused about IAX and SIP, are those 2 diff. protocols? For example, I know IAX2 works behind NAT or firewalls right? So that users with IAX softphones or phones can register into asterisk boxes behind firewall or them been behind a firewall? Is this true? Then what is SIP, another protocol and can it go behind firewalls or do you need to configure or open certain ports? __ Anton Krall ___ 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] IAX - IAX - SIP problems
The setup: Inc SIP Call - Asterisk 1 -- IAX -- Asterisk 2 -- SIP -- phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto(SIP/XX.XX.XX.XX-0819f590, cytel-internal|3044|1) in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial(SIP/XX.XX.XX.XX-0819f590, IAX2/asterisk-alpha:[EMAIL PROTECTED]/3044|30) in new stack -- Called asterisk-alpha:[EMAIL PROTECTED]/3044 -- Call accepted by XX.XX.XX.XX (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/devasterisk/1' Asterisk 2 shows the following: (CVS-HEAD) -- Registered SIP '3044' at XX.XX.XX.XX port 1911 expires 3600 -- Saved useragent CSCO/7 for peer 3044 -- Accepting AUTHENTICATED call from XX.XX.XX.XX, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, SIP/3044,30) Dec 29 08:36:09 WARNING[1496]: chan_sip.c:1351 create_addr: No such host: 3044,30 Dec 29 08:36:09 NOTICE[1496]: app_dial.c:803 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) -- Hungup 'IAX2/[EMAIL PROTECTED]/2' So how can Asterisk2 say 'no such host' when it just registered 3044 30seconds before this call came in? I'm not using any RealTime stuff anywhere. 3044 is defined: [3044] md5secret=d2756499745e254f52a224713f1a7d91 type=friend host=dynamic nat=yes canreinvite=yes disallow=g729 context=cytel-internal any ideas? Thanks, Matthew ___ 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] iax or sip
i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. what am i missing here? randy ___ 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] iax or sip
Randy Bush wrote: iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. You may indeed loathe NATted networks, but in general they're very hard to avoid. Why would you criticize a protocol for dealing with such a thing efficiently--which, quite famously, SIP does not? Also I suspect if you spent about 2.5453 nanoseconds on a call done using *only* TCP, you would quickly have your answer wrt the use of UDP for VoIP. Do you know of a successful VoIP protocol that is entirely TCP-based? trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. I would want the PBX in the datastream in cases where multiple endpoint connections would pass through multiple IAX boxen, and in that case the trunking would save the decidedly-costly IP overhead that would be required if the endpoints were simply communicating directly--if bandwidth efficiency is a desideratum. Perhaps in your case your networks are all public-IP, running on DS3s or OC48s. In that case I don't reckon efficiency would matter much. . . . what am i missing here? ?? My guess would be experience, but that might be presumptous of me. I'll let others weigh in. Maybe I'm completely misreading this. B. ___ 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] iax or sip
On Mon, 5 Jul 2004, Randy Bush waxed: i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. Are we all supposed to guess what your needs are ? iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. I think you could argue that IAX loathes NATs, too. That's why it traverses them. That's a loathing way about it, eh ? trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. You *can* set up IAX to by-pass intermediate PBXes for direct, end-to-end communication. I think the default conf files actually ship that way. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. So future expansion with the protocol is not your concern ? what am i missing here? The answer to the question: what kind of VoIP do you want ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] iax or sip
On Mon, 5 Jul 2004 13:11:10 -0700, Randy Bush wrote: i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. If you are using an IP/AIX based termination provider then trunking makes great sense. Even in my situation with only a handlful of desktops I see enough activity to know that I'm saving bandwidth with trunking to my termination provider. Michael now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. what am i missing here? randy ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's the end of the world as we know it. I feel Fine. - R.E.M. ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] iax or sip
Randy, On 01:11 PM 7/5/2004, Randy Bush wrote: iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. SIP and H323 use in NAT'd environments is problematic. IAX provides another solution to the issue for Asterisk administrators. If you don't use a NAT, then you can ignore this feature of IAX. trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. There are many reasons to have an Asterisk box in a stream: 1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) 2. Provide features (access to PSTN, conference capability, music on hold, call parking, agents and queues. the list goes on and on) 3. Endpoints (User Agents) MAY not be able to send data streams to each other directly (firewalls or nats in the middle) And depending upon your view of things (your view might be different than the view of the IT/communications administrator of a large company), using IAX in a geographically distributed use scenario might very well be exactly what you want (use over an encrypted vpn link, etc.) now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. what am i missing here? Nothing, really. If you don't need IAX, and don't particularly like any of the features of it, then don't use it. At this time, if you don't have NAT's to deal with (or you've already convinced your signalling and media protocols to deal with them), and you don't have a need to force the streams and signalling through your asterisk systems, then 'noload = chan_iax.so' to your hearts content! -Chris ___ 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] iax or sip
hi snip trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. you call -30% (more or less, depending on the codec) in bandwidth only some bytes ? reinvite rules, especially in a geographically distributed use scenario. that could be done with iax. see the notransfer flag in iax.conf you can move the entire call away, not only the rtp stream. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. but is easy to add info to iax to carry what you need. what am i missing here? experience. btw, SIP is certainly needed 'cause of the clients... much more available than iax ones. but for server to server pov, iax is sure a better choice. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] iax or sip
Randy Bush wrote: what am i missing here? Use SIP in the LAN and IAX in the WAN. Jeremy McNamara ___ 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] IAX or SIP termination provider
I'm in Mexico an I'll like to know wish is the best IAX or SIP Termination provider. Im tring to start a small Pre-paid long distance service. Thanks Erick ___ 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] IAX vs SIP
Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0, wonder if it's a softphone's problem? I have got X-Lite to work with G.711 and GSM only, I have never been able to get it to work with iLBC or Speex.. I use iLBC over my IAX trunk and it works fine so I can only guess that there is some compatibility problem between X-Lite and Asterisk.. Later -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0, wonder if it's a softphone's problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX vs SIP
How do you set up IAX in Trunk mode? Uriel Add trunk=yes to your definition in iax.conf.. Later -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
FYI: trunking only works in IAX2 and it requires you to have a zaptel interface on both endpoints I have heard that but in my setup I only have Zaptel hardware on one side and trunking appears to work fine.. Initially I tried using ztdummy on the side which didn't have zaptel hardware but this caused the trunk to break properly, without it it works fine.. Maybe I just have a freak setup.. :) Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Hello, On 19-09 19:48, WipeOut . wrote: Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. I am wondering how setup like this could work with IAX (or any other protocol) when symmetric NATs are used. If you have two different NATs then direct connection is not possible between hosts behind those two NATs. You have to do some kind of provisioning of the NAT boxes (i.e. port forwarding). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
I am wondering how setup like this could work with IAX (or any other protocol) when symmetric NATs are used. If you have two different NATs then direct connection is not possible between hosts behind those two NATs. You have to do some kind of provisioning of the NAT boxes (i.e. port forwarding). Jan. You setup port forwarding on your each NAT's to the server behind the NAT.. If you don't have a static IP or resolvable DNS name on one of the boxes you can get it to register with the remote side.. You will have to have the NAT's public IP on at least one side static or resolvable through some form of DNS or DDNS.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ 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] IAX vs SIP
On Fri, 19 Sep 2003, WipeOut . wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. FYI: trunking only works in IAX2 and it requires you to have a zaptel interface on both endpoints James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX vs SIP
How do you set up IAX in Trunk mode? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Friday, September 19, 2003 3:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX vs SIP I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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
