Re: [asterisk-users] h extension isn't processed after call file finishes.
Matthew, I don't think I've been as clear as I'd like. I've got some fax-connected TA's that make outbound calls. The dial plan routes those calls to an AGI script that captures the fax image, the destination phone number, and creates a call file to deliver the image to the destination. The first outbound call executes the h extension when it is hung up. The second call, created by the call file, doesn't execute the h extension, even though I use the dialplan to actually route the outbound call. So, I'm ending up with statistics on the reception of the fax, but not the final delivery. Does that make more sense? Mike. On Wed, Feb 19, 2014 at 6:10 PM, Matthew Jordan wrote: > On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards > wrote: >> On Mon, 17 Feb 2014, Mike Diehl wrote: >> >>> Is there something I need to do in order to get the h extension to get >>> called? >> >> >> Would the 'g' dial() option help? >> >> "Proceed with dialplan execution at the current extension if the destination >> channel hangs up." >> >> It won't take you to h, but it may allow you to do what you need to do -- >> even if the next dialplan priority just says 'goto h.' >> > > I'm actually a bit confused about what channel(s) are executing the > 'h' extension. From the description in OP's e-mail, it sounds as if at > least one channel is dropping into the 'h' extension, and some > channels are not. Which channels are they? If it is the outbound > channel, then since that channel doesn't execute dialplan, it will > never get put into the 'h' extension, unless you use the Dial > application's 'e' option. If you want hangup logic and you're using > Asterisk 11+, you could also use a hangup handler on the outbound > channel. > > But otherwise, I would expect that the 'h' extension would always be > fired for a channel executing dialplan, so long as it is in the same > context. > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- 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] h extension isn't processed after call file finishes.
On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards wrote: > On Mon, 17 Feb 2014, Mike Diehl wrote: > >> Is there something I need to do in order to get the h extension to get >> called? > > > Would the 'g' dial() option help? > > "Proceed with dialplan execution at the current extension if the destination > channel hangs up." > > It won't take you to h, but it may allow you to do what you need to do -- > even if the next dialplan priority just says 'goto h.' > I'm actually a bit confused about what channel(s) are executing the 'h' extension. From the description in OP's e-mail, it sounds as if at least one channel is dropping into the 'h' extension, and some channels are not. Which channels are they? If it is the outbound channel, then since that channel doesn't execute dialplan, it will never get put into the 'h' extension, unless you use the Dial application's 'e' option. If you want hangup logic and you're using Asterisk 11+, you could also use a hangup handler on the outbound channel. But otherwise, I would expect that the 'h' extension would always be fired for a channel executing dialplan, so long as it is in the same context. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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] Dynamically setting from domain when calling friends
On Wed, Feb 19, 2014 at 12:12 PM, Torbjörn Abrahamsson wrote: > Thank you very much. I will try this! It seems to be what I'm looking for. > > I'm in most cases working with 1.2 asterisks, so I'm not up to date on newer > features. My current project however needed a newer version. I tried some > googleing, but I did not find these variables. Glad to help! Wow.. 1.2 ! Most are using 1.8 or 11 these days, so it is good to be aware of that when seeking help and Googeling. The 1.8 branch is the oldest supported version at the moment. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- 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] Dynamically setting from domain when calling friends
Thank you very much. I will try this! It seems to be what I'm looking for.
I'm in most cases working with 1.2 asterisks, so I'm not up to date on newer
features. My current project however needed a newer version. I tried some
googleing, but I did not find these variables.
Thanks,
Torbjörn Abrahamsson
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rusty Newton
Sent: den 19 februari 2014 16:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dynamically setting from domain when calling
friends
Actually SIPFROMDOMAIN was documented here:
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
, but SIPFROMUSER was not. They are now both there! :)
On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton wrote:
> On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
> wrote:
>> I have a problem where I would like to be able to send an arbitrary SIP
>> domain when sending a call to a registered friend. By default the from
>> domain is set to the IP of the Asterisk server, but I would like to set it
>> to something else.
>> The case is that when a call from a foreign domain comes in to the Asterisk,
>> it will connect it to the callee (but with the domain changed). When the
>> callee wants to make a redial from call history, the domain will not be
>> correct.
>> I could probably do something with the fromdomain setting of the friend, but
>> I would like it to be dynamic, ie not having to update the friend definition
>> every time a different domain is used.
>> I understand that I would need to use outbound proxy in the client to
>> prevent it from dialing the domain directly.
>> Is this possible? Any alternatives?
>
> I'm a little confused about what you want to do, however I'll throw
> some information at you in hopes that it will help out.
>
> I did a little research and found that you can set the outbound From
> header domain and From header user through two channel variables:
> SIPFROMDOMAIN, SIPFROMUSER
>
> They are sparsely documented, but there is an example in extensions.conf
>
> same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
> same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)
> ; check if we set the FREENUMDOMAIN global variable in [global]
> same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
> ;if we did set it, then we'll use it for our outbound dialing
> domain
>
> It looks like they were added in 1.6.2.X of Asterisk, so if you are
> using 1.8.X or above, you should have them.
>
> On your inbound call, you could use the function SIP_HEADER[1] to
> gather the domain and store it for later use when you want to set it
> on the outbound call. Though I'm not sure how you could tell that the
> call was a redial.
>
> [1]:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
>
> I'm assuming when your SIP client redials that it calls through
> Asterisk and is not dialing the previously caller directly.
>
> Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
> SIPFROMUSER on the wiki*
>
>
> --
> Rusty Newton
> Digium, Inc. | Community Support Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct: +1 256 428 6200
>
> Check us out at: http://digium.com & http://asterisk.org
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
--
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[asterisk-users] Asterisk as a client: can I get the remote SIP server to ignore rport?
Hi list, I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a client. I'm trying to SIP REGISTER with a remote SIP provider. The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of the VM performs static NAT from the RFC IP address to a dedicated public IP address, however, they are rewriting ports at will. That's the problem. Here's an excerpt from tcpdump: IP 192.168.1.2.5060 > my.provider.com.5060: UDP, length 411 REGISTER sip:my.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK25a36d76 Max-Forwards: 70 From: ;tag=as762d7322 To: Call-ID: 778c50f84e80a9db60dcd35a2f8a1498@127.0.0.1 CSeq: 228 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 12.0.0 Expires: 120 Contact: Content-Length: 0 Then the remote SIP provider answers of course with "401 Unauthorized" but that reply never makes it to Asterisk, because it doesn't come in on port 5060, where it actually originated on the VM, but on a random port that the VM hosting providers' NAT router rewrote to, in the below case port 63664. And the remote SIP provider tries to send the reply back on that random port. Note MY.PUBLIC.IP.ADDRESS and rport below: IP my.provider.com.5060 > 192.168.1.2.63664: UDP, length 534 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK25a36d76;received=MY.PUBLIC.IP.ADDRESS;rport=63664 From: ;tag=as762d7322 To: ;tag=as45cffa11 Call-ID: 778c50f84e80a9db60dcd35a2f8a1498@127.0.0.1 CSeq: 228 REGISTER Server: FPBX-2.10.0(1.8.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d46fec6" Content-Length: 0 I'm thinking the answer is "no", but is there any option how I can get the remote SIP provider to answer me on port 5060? Without having them to change anything in their config. Thank you! Markus -- _ -- 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] PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP
A reboot of the system after hours appears to have solved the issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg Sent: Wednesday, February 19, 2014 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP Eric! The pcap trace seems to contain only idle data, and there is nothing unusual. As far as I know, PRI always uses the static TEI value of 0, as there is only a single "terminal equipment (TE). The TEI assignment procedure is only relevant if there is a BRI connection on a so called "S0 bus" and the TE operates in point-to-multipoint mode. Then the automatically assigned TEI is within the range 64 to 126. Sangoma ISDN products using DAHDI/libpri do not have any problem with this in any mode. There are always spurious MDL (related to supervisory frames, or so) messages, but they have no impact on actual calls. If there many calls on ISDN lines and you need to restart Asterisk/DAHDI/Wanpipe, then it is likely that the system is in an inconsistent state. Some phones ring, but cannot be answered, or answered and the other side cannot hear the other side, and stuff like that. I don't see right now how this relates to your problem, but you could try to capture the SETUP (followed by CALL PROCEEDING, ALERTING, CONNECT, STATUS, ...) messages from in- and outbound calls and log what happens. jg -- _ -- 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] PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP
Eric! The pcap trace seems to contain only idle data, and there is nothing unusual. As far as I know, PRI always uses the static TEI value of 0, as there is only a single "terminal equipment (TE). The TEI assignment procedure is only relevant if there is a BRI connection on a so called "S0 bus" and the TE operates in point-to-multipoint mode. Then the automatically assigned TEI is within the range 64 to 126. Sangoma ISDN products using DAHDI/libpri do not have any problem with this in any mode. There are always spurious MDL (related to supervisory frames, or so) messages, but they have no impact on actual calls. If there many calls on ISDN lines and you need to restart Asterisk/DAHDI/Wanpipe, then it is likely that the system is in an inconsistent state. Some phones ring, but cannot be answered, or answered and the other side cannot hear the other side, and stuff like that. I don't see right now how this relates to your problem, but you could try to capture the SETUP (followed by CALL PROCEEDING, ALERTING, CONNECT, STATUS, ...) messages from in- and outbound calls and log what happens. jg -- _ -- 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] Dynamically setting from domain when calling friends
Actually SIPFROMDOMAIN was documented here:
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
, but SIPFROMUSER was not. They are now both there! :)
On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton wrote:
> On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
> wrote:
>> I have a problem where I would like to be able to send an arbitrary SIP
>> domain when sending a call to a registered friend. By default the from
>> domain is set to the IP of the Asterisk server, but I would like to set it
>> to something else.
>> The case is that when a call from a foreign domain comes in to the Asterisk,
>> it will connect it to the callee (but with the domain changed). When the
>> callee wants to make a redial from call history, the domain will not be
>> correct.
>> I could probably do something with the fromdomain setting of the friend, but
>> I would like it to be dynamic, ie not having to update the friend definition
>> every time a different domain is used.
>> I understand that I would need to use outbound proxy in the client to
>> prevent it from dialing the domain directly.
>> Is this possible? Any alternatives?
>
> I'm a little confused about what you want to do, however I'll throw
> some information at you in hopes that it will help out.
>
> I did a little research and found that you can set the outbound From
> header domain and From header user through two channel variables:
> SIPFROMDOMAIN, SIPFROMUSER
>
> They are sparsely documented, but there is an example in extensions.conf
>
> same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
> same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)
> ; check if we set the FREENUMDOMAIN global variable in [global]
> same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
> ;if we did set it, then we'll use it for our outbound dialing
> domain
>
> It looks like they were added in 1.6.2.X of Asterisk, so if you are
> using 1.8.X or above, you should have them.
>
> On your inbound call, you could use the function SIP_HEADER[1] to
> gather the domain and store it for later use when you want to set it
> on the outbound call. Though I'm not sure how you could tell that the
> call was a redial.
>
> [1]:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
>
> I'm assuming when your SIP client redials that it calls through
> Asterisk and is not dialing the previously caller directly.
>
> Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
> SIPFROMUSER on the wiki*
>
>
> --
> Rusty Newton
> Digium, Inc. | Community Support Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct: +1 256 428 6200
>
> Check us out at: http://digium.com & http://asterisk.org
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
--
_
-- 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:
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Re: [asterisk-users] Dynamically setting from domain when calling friends
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
wrote:
> I have a problem where I would like to be able to send an arbitrary SIP
> domain when sending a call to a registered friend. By default the from
> domain is set to the IP of the Asterisk server, but I would like to set it
> to something else.
> The case is that when a call from a foreign domain comes in to the Asterisk,
> it will connect it to the callee (but with the domain changed). When the
> callee wants to make a redial from call history, the domain will not be
> correct.
> I could probably do something with the fromdomain setting of the friend, but
> I would like it to be dynamic, ie not having to update the friend definition
> every time a different domain is used.
> I understand that I would need to use outbound proxy in the client to
> prevent it from dialing the domain directly.
> Is this possible? Any alternatives?
I'm a little confused about what you want to do, however I'll throw
some information at you in hopes that it will help out.
I did a little research and found that you can set the outbound From
header domain and From header user through two channel variables:
SIPFROMDOMAIN, SIPFROMUSER
They are sparsely documented, but there is an example in extensions.conf
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)
; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
;if we did set it, then we'll use it for our outbound dialing
domain
It looks like they were added in 1.6.2.X of Asterisk, so if you are
using 1.8.X or above, you should have them.
On your inbound call, you could use the function SIP_HEADER[1] to
gather the domain and store it for later use when you want to set it
on the outbound call. Though I'm not sure how you could tell that the
call was a redial.
[1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
I'm assuming when your SIP client redials that it calls through
Asterisk and is not dialing the previously caller directly.
Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
SIPFROMUSER on the wiki*
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP
It seems a layer 2 problem, should check about references point (network or terminal equipment), it assume your Asterisk box is connected to ISDN PSTN provided, just in case check at you side if all related configuration files are configured as signalling=pri_cpe (Card config, wan_cfg, chan_dahdi.conf), PSTN side if network configuration or in service mode, should both side work and debug in the same time. Mc GRATH Ricardo E-Mail mcgra...@mail2web.com -- _ -- 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] Asterisk NAT
Anyway Thank you guys. ;-) On Wed, Feb 19, 2014 at 12:25 PM, A J Stiles wrote: > On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: > > Hello, a few days ago I sent a question: > > > > > http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html > > > > but no one answered me! I just want to know is it possible or not? > > There is a bit of a tendency on this list to ignore questions that have > been > answered before. It's disconcerting at first, but remember: *you* are the > stereotype tourist here, and *not repeating oneself* is a part of the > natives' > culture. It is not exactly rudeness, per se, even though it might look > that > way; just an aversion to saying the same thing twice. > > No answer on the list probably just means the question was answered > before; so > your best bet is to search the mailing list archives and the wiki at > http://voip-info.org > Eventually, you will have been yomping around in Tech Land for long enough > to > graduate from "ignorant tourist" to "seasoned traveller" -- and then you > get > to ignore noob questions yourself. Or set yourself up as a tour guide, if > you > feel that way inclined :) > > -- > AJS > > Answers come *after* questions. > > -- > _ > -- 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] Asterisk NAT
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: > Hello, a few days ago I sent a question: > > http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html > > but no one answered me! I just want to know is it possible or not? There is a bit of a tendency on this list to ignore questions that have been answered before. It's disconcerting at first, but remember: *you* are the stereotype tourist here, and *not repeating oneself* is a part of the natives' culture. It is not exactly rudeness, per se, even though it might look that way; just an aversion to saying the same thing twice. No answer on the list probably just means the question was answered before; so your best bet is to search the mailing list archives and the wiki at http://voip-info.org Eventually, you will have been yomping around in Tech Land for long enough to graduate from "ignorant tourist" to "seasoned traveller" -- and then you get to ignore noob questions yourself. Or set yourself up as a tour guide, if you feel that way inclined :) -- AJS Answers come *after* questions. -- _ -- 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] Asterisk NAT
On 19/2/14 4:53 am, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? I can't help on the "can Asterisk detect they're behind the same NAT" part of the question, but I would caution you that an assumption that 'each NAT box has a single external IP' is risky - especially if you have to deal with the possibility of double-NAT and other such evilness (and it's hard to avoid sometimes - how many non-tech people go and buy a wireless router to 'extend their WiFi' rather than an access point, or don't know how to switch said router into AP-only mode). You also have to consider users who have multiple LANs (which might not necessarily be able to route between themselves) behind a single external IP: this one's quite common in shared/managed office environments - one external IP and several RFC1918 VLANs internally, with no routing between them. So in summary, unless you have a considerable level of control over your endpoints such that you can be sure these (and no doubt other) scenarios don't apply, it's probably safest to send RTP traffic through Asterisk regardless, otherwise you're potentially opening up a support nightmare for yourself. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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
