Re: [asterisk-users] func_curl.so Error on load
On Sun, Apr 20, 2008 at 07:37:32PM -0700, Chris Brentano wrote: When I ran ./configure, which completed successfully, I noticed that it complained about the PKG_CONFIG_PATH and not being able to find libcurl: (lines omitted) ... checking for curl-config... /usr/bin/curl-config Package libcurl was not found in the pkg-config search path. Perhaps you should add the directory containing `libcurl.pc' to the PKG_CONFIG_PATH environment variable No package 'libcurl' found Sounds like autoconf not looking good enough. Or a bug in the package you used. ... Which, was ridiculous that it finished ./configure and didn't error out on the spot, since without this small piece of the puzzle Asterisk would not run. It will: libcurl is not required for building Asterisk. Generally for most of the optional libraries, the confogure script of Asterisk will silently fail if they are not installed. I don't think you want to have to install snmp, unixodbc, openh323, libpri, libvpb and whatever just to get Asterisk built. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TDM410P Cards
As recommened I got the new firmware for my echo cancellers and it solved hte problem with the agressive echo cancelling causing half duplex audio. I have to say, so far these cards are far superior to the previous models. The sound quality is hugely improved (enough to really notice which is alot) and the echo canceller works way better than the software ones. My system seems to like these cards must better too, no more irq issues so far. So I will now be using digium cards once again, i stopped for a while after the issues i reported here caused me lots of headaches. I am really glad digium got these cards fixed because they have a much better price point than the competition. I will report back after a months worth of usage. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_curl.so Error on load
Generally I'd agree. But it could at least more adequately notify the user, even if they are compiling on a different system than where it will be running on. It just seems that in most cases people will be compiling on the system they will be installing on. This is what they teach at the Asterisk Bootcamp, fwiw. Tzafrir Cohen wrote: It will: libcurl is not required for building Asterisk. Generally for most of the optional libraries, the confogure script of Asterisk will silently fail if they are not installed. I don't think you want to have to install snmp, unixodbc, openh323, libpri, libvpb and whatever just to get Asterisk built. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic Possiblity Question.
Hi all, i have a basic question on asterisk.The below is my scenerao. I have my sales offices around the globe.Theyare all connected with Speed Internet connection.I don't mind installing 1 asterisk box in each site.i don't mind using IP phone.i just wanted to call them for free at the cost of existing internet connectionwe have at each site.All the asterisk box will be connected with TCP/IP with one of it's NIC card having a WAN connectivity.is it possible with asterisk.Please let me know.Thanx _ Going green? See the top 12 foods to eat organic. http://green.msn.com/galleries/photos/photos.aspx?gid=164ocid=T003MSN51N1653A___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip channel - detect ringing (nvlinedetect??)
Hello ppl, Is there any other way to detect states like Ringing on SIP channels on Asterisk? Nvlinedetect is one way, but it seems to have disappeared from the face of the earth! Any pointers or does anyone have the code for NV* features? Thanks in advance - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] API Originate - action on reject/busy/congestion
Hello ppl, I am using the Astman API Originate command to initiate a call to a user. On connect of the user, I dial another user to bridge the call between the two. I am using the Async option with the Originate command, as I don't want to use Astman proxy yet. Is there any way to invoke a script, etc if the first user doesn't pick up the call/rejects it or we get a congestion on that channel? TiA, - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Visualization (Extensions.conf or Dialplan Show)
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote: will this do? http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0-0.1.0-2.html btw, it has (almost) nothing to do with trixbox Considering it takes the data from a mysql table called asterisk and hard-wires the default FreePBX (or is it TrixBox CE) password for that table, I'd say it has everything to do with FreePBX. Thanks, but yeah, we had found this already too. It's too tied to FreePBX for our use, we want essentially the same thing but more like asterisk -rx dialplan show | fancygrapherscript.ext :) We're going to look at the graphiz stuff this has incorporated and see if we can do some really crazy regexes to accomplish this.. unless there is a way to make asterisk export the dialplan in xml format or some form of structured format that anyone knows of.. ? thanks, matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] imaps - voicemail
I added the following to the voicemail config files: imapserver=imap.gmail.com imapport=993 and imapuser=user|imapsecret=pass to the mailbox details. However that just causes asterisk to hang as soon as i try to use that extensions... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] re-invite (bypass asterisk) post call establishment
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote: Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. You don't say what you've tried already, but as long as canreinvite=yes is set against the SIP peer, the RTP stream should be redirected once the connection is open. As far as DTMF to dial an extension at the remote end, have you looked at the D() parameter to the Dial command? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic Possiblity Question.
Sure - read up on IAX for a few good points. PaulH rupak shrestha [EMAIL PROTECTED] wrote: Hi all, i have a basic question on asterisk.The below is my scenerao. I have my sales offices around the globe.Theyare all connected with Speed Internet connection.I don't mind installing 1 asterisk box in each site.i don't mind using IP phone.i just wanted to call them for free at the cost of existing internet connectionwe have at each site.All the asterisk box will be connected with TCP/IP with one of it's NIC card having a WAN connectivity.is it possible with asterisk.Please let me know.Thanx _ Going green? See the top 12 foods to eat organic. http://green.msn.com/galleries/photos/photos.aspx?gid=164ocid=T003MSN51N 1653A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX using Outbound proxy
On 18/04/2008, Rosa De Santis [EMAIL PROTECTED] wrote: Hi all. Please, how can I configure an Asterisk PBX using an outbound proxy (that resolve NAT Traversal) I'm trying using the outboundproxy and outboundproxyport values in sip.conf but the PBX don't get registered on the outbound proxy side. I'm using SER + Asterisk with Jasomi outbound proxy solution, and I want the PBX to have a SIP trunk, but in SER i see the pbx sip user registered as [EMAIL PROTECTED] , not the SER ip. Any ideas please? Thanks in advance, Rosa. What register = line are you using? Take a look at: http://bugs.digium.com/view.php?id=12474 Perhaps it will help? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound PRI ISDN 30 problems
On 20/04/2008, robert boardman [EMAIL PROTECTED] wrote: Hi All I'm having problems with outboud ISDN calls, They setup OK , and ring the other end OK, but when the call is answered I get a disconnect cuase 17 with an error message in the console of [Apr 15 08:06:13] DEBUG[4361] chan_zap.c: Found empty available channel 0/31 [Apr 15 08:06:13] VERBOSE[4601] logger.c: -- Starting simple switch on 'Zap/62-1' [Apr 15 08:06:13] VERBOSE[4361] logger.c: -- Accepting overlap call from '12345678901' to '0797' on channel 0/31, span 2 [Apr 15 08:06:13] VERBOSE[4361] logger.c: -- Channel 0/31, span 2 got hangup, cause 17 [Apr 15 08:06:13] WARNING[4601] channel.c: Unexpected control subclass '5' Any assistance would be greatly appriciated I would suggest posting your zaptel.conf and zapata.conf files, and perhaps the appropriate part of your dialplan so we can see what might be happening. As far as hangup-cause codes, look here: http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Visualization (Extensions.conf or Dialplan Show)
On Mon, Apr 21, 2008 at 5:11 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 04:19:16AM -0400, Matthew Gibson wrote:
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen
[EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote:
will this do?
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0-0.1.0-2.html
btw, it has (almost) nothing to do with trixbox
Considering it takes the data from a mysql table called asterisk and
hard-wires the default FreePBX (or is it TrixBox CE) password for that
table, I'd say it has everything to do with FreePBX.
Thanks, but yeah, we had found this already too. It's too tied to
FreePBX
for our use, we want essentially the same thing but more like
asterisk -rx dialplan show | fancygrapherscript.ext
Here's a quick hack. Just looked at the dot man page. Did't even test
that it is actually valid. But it is probably a good start.
asterisk -rx 'dialplan show' | \
awk -F' '
BEGIN {printf digraph dialplan {\n;};
/^\[ Context/ {context=$2};
/^ Include =/ {printf \t%s - %s\n,context,$2};
END {printf }\n}
'
Only graphs simple inclusions between contexts.
BTW:
asterisk -rx | less
Behaves really strange. no scrolling possible. But I don't see anything
strange in a hexdump.
Awesome. Thanks!
Will post updates as we progress.
Thanks,
Matt
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Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. Ok. So, since the bug we logged was closed and these tests weren't registered along with it, when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? Thanks in advance, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: UMA in UK, any use?
Hi,sorry for off topic post, struggling to find any information on UMA in the UK. I have a Blackberry 8320 phone with wi-fi and UMA capability, its actually an unlocked Orange branded phone. T-Mobile don't support UMA in the UK, is it possible to do anything else with the UMA feature of this phone? Or, is it totally locked to your network provider? Any possible way of hacking it to work as some kind of voip client to work on one's own implementation of UMA, if such a thing even exists? :) Thanks Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] re-Invite post call establishment (for RTP bypass)
Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] re-invite (bypass asterisk) post call establishment
Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] re-Invite post call establishment (for RTP bypass)
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote: Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. Is there an echo? ;-) I answered this an hour ago. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Visualization (Extensions.conf or Dialplan Show)
On Mon, Apr 21, 2008 at 04:19:16AM -0400, Matthew Gibson wrote:
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote:
will this do?
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0-0.1.0-2.html
btw, it has (almost) nothing to do with trixbox
Considering it takes the data from a mysql table called asterisk and
hard-wires the default FreePBX (or is it TrixBox CE) password for that
table, I'd say it has everything to do with FreePBX.
Thanks, but yeah, we had found this already too. It's too tied to FreePBX
for our use, we want essentially the same thing but more like
asterisk -rx dialplan show | fancygrapherscript.ext
Here's a quick hack. Just looked at the dot man page. Did't even test
that it is actually valid. But it is probably a good start.
asterisk -rx 'dialplan show' | \
awk -F' '
BEGIN {printf digraph dialplan {\n;};
/^\[ Context/ {context=$2};
/^ Include =/ {printf \t%s - %s\n,context,$2};
END {printf }\n}
'
Only graphs simple inclusions between contexts.
BTW:
asterisk -rx | less
Behaves really strange. no scrolling possible. But I don't see anything
strange in a hexdump.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
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[asterisk-users] Asterisk Jingle-SIP GW Question
Dear All I am using gtalk features with my own XMPP server OpenFire I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls from clients registered on my XMPP server to SIP devices by calling the xmpp accounts registered as clients on asterisk. So far so good. So if I want to call sip:1000 I call the xmpp account that is bound to that account in extensions.conf. However what do I have to do to make this work with PSTN numbers. I can just setup an entry + extensions for each pstn number I want to call. I know that I can parse the incoming number and send it to the PSTN with sip, however with jingle the number must be online already since jingle is presence based. So I must have a registered client for each number I want to call in the following format XMPP --- SIP 1000 To Call 1000 //sip extension 1001 To Call 15461315461 //pstn num 1002 To Call 46456543213 //cell phone num So in essence I need to have one entry in jabber.conf per number, is there something dynamic that can be done ? Thanks asterisk-users@lists.digium.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UPDATED Asterisk Jingle Extensions.conf
Dear All I am using gtalk features with my own XMPP server OpenFire I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls from clients registered on my XMPP server to SIP devices by calling the xmpp accounts registered as clients on asterisk. I have sent a previous email with a problem that I solved by using component mode. In this mode the asterisk server acts as a subdomain. So I can call [EMAIL PROTECTED], [EMAIL PROTECTED] My current extension file looks as follows: [google-in] exten = s,1,NoOp( Call from XMPP) exten = s,n,Set(CALLERID(name)=From XMPP Server) exten = s,n,Dial(SIP/1234) However I want it to call the number in dialed initially I.e 1000 or 1001 etc etc etc. Any way to do this parsing using Asterisk ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP stats
Hello, Is here an easy way to get RTCP Stats in channel variables after the call ends? Or source should be edited to accomplish this? I would like to know this before developing this feature. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: UMA in UK, any use?
On Mon, Apr 21, 2008 at 11:02:13AM +0100, Mike Dent wrote: sorry for off topic post, struggling to find any information on UMA in the UK. I have a Blackberry 8320 phone with wi-fi and UMA capability, its actually an unlocked Orange branded phone. T-Mobile don't support UMA in the UK, is it possible to do anything else with the UMA feature of this phone? Or, is it totally locked to your network provider? Any possible way of hacking it to work as some kind of voip client to work on one's own implementation of UMA, if such a thing even exists? :) UMA is completely tied to the operators. It requires back-end technology to transfer the call from GSM/3G to WiFi or vice versa. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Click-to-talk (Java application)
Hi! I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Visualization (Extensions.conf orDialplan Show)
I've been working on a visualization tool that would be totally open
source and I'd share it now if I wasn't embarassed by how quickly and
shoddily I assembled it. Here's an example of the sample extensions.conf
file that was part of 1.4.19:
http://www.mbs3.org/extensions-conf-sample.jpg
http://www.mbs3.org/extensions-conf-sample.jpg
Right now, it doesn't handle any of the quoting escapes that Asterisk
allows, and it shows anything that is an include or a Goto. Eventually I
think it might be neat to just graph all of the dialplan's structure,
kind of like this site explains for web sites:
http://www.aharef.info/2006/05/websites_as_graphs.htm
More soon :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Gibson
Sent: Monday, April 21, 2008 5:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan Visualization
(Extensions.conf orDialplan Show)
On Mon, Apr 21, 2008 at 5:11 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Mon, Apr 21, 2008 at 04:19:16AM -0400, Matthew Gibson
wrote:
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen
[EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe
Brevda wrote:
will this do?
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0
-0.1.0-2.html
btw, it has (almost) nothing to do with trixbox
Considering it takes the data from a mysql table
called asterisk and
hard-wires the default FreePBX (or is it TrixBox CE)
password for that
table, I'd say it has everything to do with FreePBX.
Thanks, but yeah, we had found this already too. It's
too tied to FreePBX
for our use, we want essentially the same thing but
more like
asterisk -rx dialplan show |
fancygrapherscript.ext
Here's a quick hack. Just looked at the dot man page.
Did't even test
that it is actually valid. But it is probably a good
start.
asterisk -rx 'dialplan show' | \
awk -F' '
BEGIN {printf digraph dialplan {\n;};
/^\[ Context/ {context=$2};
/^ Include =/ {printf \t%s - %s\n,context,$2};
END {printf }\n}
'
Only graphs simple inclusions between contexts.
BTW:
asterisk -rx | less
Behaves really strange. no scrolling possible. But I
don't see anything
strange in a hexdump.
Awesome. Thanks!
Will post updates as we progress.
Thanks,
Matt
___
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Re: [asterisk-users] Click-to-talk (Java application)
On 21 Apr 2008, at 14:31, equis software wrote: Hi! I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? Yep. We can help on a commercial basis. Contact me off-list if you are interested. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Quality Voice in a Asterisk-E1-Unicall
The E1 use ALAW, if you want to avoid trans-coding use ALAW in your phones as well. In any call you have 2 call legs, callee and caller, try to isolate the problem and determine if the audio is really coming that bad from the E1, you can use ztmonitor to hook into the E1 and listen to the audio. If the audio you get using ztmonitor is deficient then you know it has nothing to do with trans-coding or the codec you use in your phones. Is the Digium card missing interrupts? (zttool will tell you so) Moisés Silva On Sun, Apr 20, 2008 at 3:23 PM, Ruben Zamora [EMAIL PROTECTED] wrote: I Have with Asterisk- Unicall - E1 (MFC/R2). Days before a install a Digium Card TE122P with hardware echo cancelation, these because a had a echo in some in and out calls. I replaced the card. I no more echo but in my conversation the voice start to doing things. Like after a minutes i start hearing the voice cut. or the cant hearme.. I remove in the zaptel.conf the echotraining.I dont know if i really need to do these changes in the unicall.conf.??? In my Asterisk am using GXP2020 Grandstream what is better ulaw,alaw,g729??? I apreciate any help. Thanks Ruben ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over TCP
Hello guys, I thought it would be neat if we had a SIP client for Asterisk working in Adobe Flash, but as far as I know, Flash only supports TCP. I know that Asterisk (at least v1.6) can handle SIP communication over TCP, but I was wondering is there a possibility to route audio stream over TCP too? Regards, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click-to-talk (Java application)
Thanks, I´m interested in non comercial solutions. On Mon, Apr 21, 2008 at 11:00 AM, Tim Panton [EMAIL PROTECTED] wrote: On 21 Apr 2008, at 14:31, equis software wrote: Hi! I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? Yep. We can help on a commercial basis. Contact me off-list if you are interested. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. Ok. So, since the bug we logged was closed and these tests weren't registered along with it, when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? It should be in the next release. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor not merging calls
I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record
all incoming calls. One of the box that have Asterisk 1.4.18 is properly
merging calls and the other box that has Asterisk 1.4.15 is recording the calls
but not merging them, I have made sure that SOX is installed on the box.
Here is the Dialplan of both the machines :
exten = 1234,1,Answer()
exten = 1234,2,Monitor(gsm,/recordings)/${UNIQUEID},m)
Do I have to upgrade and check or is their some other thing I can check?
Regards,
Sanjay Rajdev
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Re: [asterisk-users] Monitor not merging calls
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote: One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. It might be worth giving the MixMonitor() application a try instead. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click-to-talk (Java application)
On Mon, 2008-04-21 at 10:31 -0300, equis software wrote: I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? You can try with jiax: http://www.hem.za.org/jiaxclient/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore, but you can use sox -m and I think asterisk should be changed to use that instead. on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote: One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. It might be worth giving the MixMonitor() application a try instead. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9
As long as you renamed it to indications.conf when you copied it then it should be working after you do a reload of Asterisk. If it's not working then the problem was not indications.conf aby azid wrote: Hi Eric, i copy the indications.conf.sample from the asterisk source and paste it in the /etc/asterisk directory. I reloaded asterisk and still the message appear when i sent call to Quintum. Am I doing it right? cheers, Aby Azid On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling [EMAIL PROTECTED] wrote: Use the indications.conf.sample that comes with the Asterisk source. aby azid wrote: Thank you for replying, How would i know, whether i have the valid indicitions.conf ? On Sun, Apr 20, 2008 at 8:47 PM, Eric Wieling [EMAIL PROTECTED] wrote: Make sure you have a valid /etc/asterisk/indications.conf aby azid wrote: Hi, this is my first ever post, would appreciate if anyone can explain it to me this status message: *[Apr 20 19:12:31] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 20 19:12:31] WARNING[759]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/quintum_kl-0940c570' [Apr 20 19:12:32] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 [Apr 20 19:12:32] WARNING[759]: channel.c:2390 ast_indicate_data: Unable to handle indication 9 for 'SIP/quintum_kl-0940c570' *this happens when I sent call to my quintum gateway server, the status appears as soon as the call get connected. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
John, Is their something that I can change on my side to get this working ? Jared, I thought MixMonitor() was for Queue, Can you let me know how to use it? Thanking you for replying. Regards, Sanjay Rajdev - Original Message - From: John covici [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Monitor not merging calls Newer version of sox don't seem to have soxmix anymore, but you can use sox -m and I think asterisk should be changed to use that instead. on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote: One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. It might be worth giving the MixMonitor() application a try instead. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
http://www.voip-info.org/wiki/view/MixMonitor On Mon, Apr 21, 2008 at 7:43 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: John, Is their something that I can change on my side to get this working? Jared, I thought MixMonitor() was for Queue, Can you let me know how to use it? Thanking you for replying. Regards, Sanjay Rajdev - Original Message - From: John covici [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Monitor not merging calls Newer version of sox don't seem to have soxmix anymore, but you can use sox -m and I think asterisk should be changed to use that instead. on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote: One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. It might be worth giving the MixMonitor() application a try instead. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic Possiblity Question.
I would look at setting up OpenVPN on each of the Asterisk boxen and running SIP between them. I have read that IAX2 is much better now, but I have had many major voice quality issues with it. With OpenVPN, all Asterisk boxen appear to each other as being on the same subnet. This gives you ease in using SIP and added security on many different facets. Thanks, Steve Totaro On 4/21/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Sure - read up on IAX for a few good points. PaulH rupak shrestha [EMAIL PROTECTED] wrote: Hi all, i have a basic question on asterisk.The below is my scenerao. I have my sales offices around the globe.Theyare all connected with Speed Internet connection.I don't mind installing 1 asterisk box in each site.i don't mind using IP phone.i just wanted to call them for free at the cost of existing internet connectionwe have at each site.All the asterisk box will be connected with TCP/IP with one of it's NIC card having a WAN connectivity.is it possible with asterisk.Please let me know.Thanx _ Going green? See the top 12 foods to eat organic. http://green.msn.com/galleries/photos/photos.aspx?gid=164ocid=T003MSN51N 1653A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATED Asterisk Jingle Extensions.conf
Hi Ali, I have sent a previous email with a problem that I solved by using component mode. In this mode the asterisk server acts as a subdomain. So I can call [EMAIL PROTECTED], [EMAIL PROTECTED] That's a nice way of using Asterisk's component capability. Which XMPP/Jingle client are you using? However I want it to call the number in dialed initially I.e 1000 or 1001 etc etc etc. Any way to do this parsing using Asterisk ? If your XMPP/Jingle client can send DTMF, you can use Asterisk's DISA application that will collect the entered digits and place a new call : http://www.voip-info.org/wiki-Asterisk+cmd+DISA If your XMPP/Jingle client cannot send DTMF, then please open a feature request on the bug tracker : http://bugs.digium.com/, along with an Asterisk's debug output and detailed description of the feature. Cheers, Philippe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone notification?
Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dual Interface config
I'm looking for some configuration help. I'm currently running Asterisk 1.4 on Centos 5. I have a server that has two network cards, the first card is a public ip that does sip trunking to our sip provider. The second network card is an internal ip that is a seperate voice vlan. The problem that I'm having is that when I dial out via our sip trunk, it appears that asterisk is reinviting the handset and our sip trunk to talk direct. This won't work because our sip provider will only accept traffic from our public facing ip. I thought if I set caninvite=no and reinvite=no this would cause asterisk to continue processing the media. Is that not the case? I've scraped through what documentation I can find and googled but the only additional info I could find was to set the externip=MYPUBLICIP. Can anyone with a similar setup help point me in the right direction? Thanks, Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor v/s MixMonitor
What is good for recording all the incoming and outgoing calls, Monitor() or MixMonitor(). Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone notification?
AnDY wrote: Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), If I understand you correctly, you'll be interested in this bug: http://bugs.digium.com/view.php?id=8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
I changed the codeof the monitor app to use sox -m instead of soxmix
which I no longer have. Mixmonitor would work as well, but the
one-touch recording was set to the other, so I am using that.
on Monday 04/21/2008 Sanjay Rajdev([EMAIL PROTECTED]) wrote
John,
Is their something that I can change on my side to get this working ?
Jared,
I thought MixMonitor() was for Queue, Can you let me know how to use it?
Thanking you for replying.
Regards,
Sanjay Rajdev
- Original Message -
From: John covici [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai,
New Delhi
Subject: Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore, but you can
use sox -m and I think asterisk should be changed to use that instead.
on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote:
One of the box that have Asterisk 1.4.18 is properly merging calls and
the other box that has Asterisk 1.4.15 is recording the calls but not
merging them, I have made sure that SOX is installed on the box.
It might be worth giving the MixMonitor() application a try instead. :-)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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How do
you spend it?
John Covici
[EMAIL PROTECTED]
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htmlheadstyle type='text/css'p { margin: 0; }/stylestyle
type='text/css'body { font-family: 'Times New Roman'; font-size: 12pt;
color: #00}/style/headbodyJohn,brIs their something that I can
change on my side to get this working?brbrJared,brI thought
MixMonitor() was for Queue, Can you let me know how to use
it?brbrThanking you for replying.brbrRegards,brSanjay
Rajdevbrbr- Original Message -brFrom: John covici lt;[EMAIL
PROTECTED]gt;brTo: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt;brSent:
Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai, New
DelhibrSubject: Re: [asterisk-users] Monitor not merging
callsbrbrNewer version of sox don't seem to have soxmix anymore, but you
canbruse sox -m and I think asterisk should be changed to use that
instead.brbron Monday 04/21/2008 Jared Smith([EMAIL PROTECTED])
wrotebrnbsp;gt; On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev
wrote:brnbsp;gt; gt; One of the box that have Asterisk 1.4.18 is
properly merging calls andbrnbsp;gt; gt; the other box that has
Asterisk 1.4.15 is recording the calls but notbrnbsp;gt; gt; merging
them, I have made sure that SOX is installed on the box. brnbsp;gt;
brnbsp;gt; It might be worth giving the MixMonitor() application a try
instead. :-)brnbsp;gt; brnbsp;gt; brnbsp;gt; -- brnbsp;gt;
Jared Smithbrnbsp;gt; Community Relations Managerbrnbsp;gt; Digium,
Inc.brnbsp;gt; brnbsp;gt; brnbsp;gt;
___brnbsp;gt; -- Bandwidth
and Colocation Provided by http://www.api-digital.com --brnbsp;gt;
brnbsp;gt; asterisk-users mailing listbrnbsp;gt; To UNSUBSCRIBE or
update options visit:brnbsp;gt; nbsp;
nbsp;http://lists.digium.com/mailman/listinfo/asterisk-usersbrbr--
brYour life is like a penny. nbsp;You're going to lose it. nbsp;The
question is:brHow dobryou spend it?brbrnbsp;nbsp; nbsp; nbsp;
nbsp; John Covicibrnbsp;nbsp; nbsp; nbsp; nbsp; [EMAIL
PROTECTED]brbr___br--
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How do
you spend it?
John Covici
[EMAIL PROTECTED]
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[asterisk-users] Switch recommendation?
This will be my first major asterisk experiment and I'm trying to choose a PoE switch for 15-24 phones. I was going to spend $400 on this: http://www.newegg.com/product/product.asp?item=N82E16833124053 but then I see this on ebay: http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZWDVWQQrdZ1QQcmdZViewItem and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the Cisco IP phone's proprietary wizardry be a problem for my flock on Linksys IP phones? Because as long as it can do vlan qos and poe I think I can scrape by for half the price, right? Thanks for reading! -- Just Hil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor v/s MixMonitor
MixMonitor. And please stop posting the same question to the list over and over. Sanjay Rajdev wrote: What is good for recording all the incoming and outgoing calls, Monitor() or MixMonitor(). Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
I am probably not too qualified to answer this question but I would go with the linksys. Fred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hilary Miller Sent: Monday, April 21, 2008 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Switch recommendation? This will be my first major asterisk experiment and I'm trying to choose a PoE switch for 15-24 phones. I was going to spend $400 on this: http://www.newegg.com/product/product.asp?item=N82E16833124053 but then I see this on ebay: http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZ WDVWQQrdZ1QQcmdZViewItem and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the Cisco IP phone's proprietary wizardry be a problem for my flock on Linksys IP phones? Because as long as it can do vlan qos and poe I think I can scrape by for half the price, right? Thanks for reading! -- Just Hil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:480cec7523281809518707! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Interface config
external ip for an internal server ? sounds too dangerouse to me. i would suggest you put the server back to local lan use a router to hold your external ip do port forwarding to internal servers. it will solve your dilema keep your server safe. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
Woah, How weird. I JUST bought this off of ebay 2 minutes ago. The exact one. This will be my first time playing with PoE. I have all cisco phones here but I'll let you know how it goes. This will be my first major asterisk experiment and I'm trying to choose a PoE switch for 15-24 phones. I was going to spend $400 on this: http://www.newegg.com/product/product.asp?item=N82E16833124053 but then I see this on ebay: http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZWDVWQQrdZ1QQcmdZViewItem and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the Cisco IP phone's proprietary wizardry be a problem for my flock on Linksys IP phones? Because as long as it can do vlan qos and poe I think I can scrape by for half the price, right? Thanks for reading! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] buying cards from pakistan
On Fri, 18 Apr 2008 11:30:46 -0400 Steve Totaro [EMAIL PROTECTED] wrote: n Fri, 18 Apr 2008 18:40:17 +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, 19 Apr 2008 15:07:47 +0100 Alan Lord [EMAIL PROTECTED] wrote: thanks to everybody; I'm happy to know no responses were hostile :-) but I will not abuse; I will try all your suggestions and I will report my results; thank you very much, giuliano curti ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable transfer on all calls
Hi folks, I have some asterisk 1.2 box with self-made billing, and I need to disable call transfer on all calls and directions. I turned it off in features.conf and there is no 'tT' option in all my Dial() commands, but users still able to transfer call using transfer function in ip of softphones (AFAIK this function uses SIP method REFER), so this transfers are hard to trace in CDR and my users can make a free call using trick with transfer:) I've googled it, but didn't find anything about my problem :( Thanks, Danila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
Hilary Miller wrote: This will be my first major asterisk experiment and I'm trying to choose a PoE switch for 15-24 phones. I was going to spend $400 on this: http://www.newegg.com/product/product.asp?item=N82E16833124053 but then I see this on ebay: http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZWDVWQQrdZ1QQcmdZViewItem and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the Cisco IP phone's proprietary wizardry be a problem for my flock on Linksys IP phones? Because as long as it can do vlan qos and poe I think I can scrape by for half the price, right? Thanks for reading! The Cisco 3524 switch doesn't support 802.3af which is what your Linksys phones are going to want. If you have just Cisco phones this would work. To have 802.3af you have to have at least a Cisco 3560 series switch. See: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00801189b5.shtml#powerover for reference ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Interface config
canreinvite=no should just work, if it doesn't then maybe you want to post parts of your SIP conf. Thanks, Steve Totaro On 4/21/08, linuxian iandsd [EMAIL PROTECTED] wrote: external ip for an internal server ? sounds too dangerouse to me. i would suggest you put the server back to local lan use a router to hold your external ip do port forwarding to internal servers. it will solve your dilema keep your server safe. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Mon, Apr 21, 2008 at 4:38 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: ...when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? It should be in the next release. Great. Thanks for your feedback. We will be waiting for it... -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
We've been very happy with the SRW224Ps we've deployed. (noisy as hell... good for either the datacentre / computer room or for installation in a noise-cancelling cabinet... but then again, are there any PoE switches that aren't ?) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone notification?
http://sipsak.org/ has enabled people to display many different things on their phones. I have yet to do this but have seen it mentioned more than once. Thanks, Steve Totaro On 4/21/08, Doug Lytle [EMAIL PROTECTED] wrote: AnDY wrote: Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), If I understand you correctly, you'll be interested in this bug: http://bugs.digium.com/view.php?id=8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis [EMAIL PROTECTED] wrote: The Cisco 3524 switch doesn't support 802.3af which is what your Linksys phones are going to want. Thank you for sharing Sean! When I saw them I felt a disturbance in the force, and now I know why! -- Just Hil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic Possiblity Question.
The answer to your question depends on the QOS you desire. If you are concerned less with call quality, and more with price, then the previous solutions are certainly adequate. If call quality is important to you, I would recommend going with a VOIP provider that has a private network, that way your calls don't get switched to the public internet. When calls are switched to the public internet, quality issues are much more likely and random. WIth a private network your calls will be of much better quality and problems are less likely. Also, I have no experience with multiple Asterisk Boxes myself, but I have heard it can be tricky to cluster them as it seems you would be likely to be doing. On Mon, Apr 21, 2008 at 1:21 PM, Steve Totaro [EMAIL PROTECTED] wrote: I would look at setting up OpenVPN on each of the Asterisk boxen and running SIP between them. I have read that IAX2 is much better now, but I have had many major voice quality issues with it. With OpenVPN, all Asterisk boxen appear to each other as being on the same subnet. This gives you ease in using SIP and added security on many different facets. Thanks, Steve Totaro On 4/21/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Sure - read up on IAX for a few good points. PaulH rupak shrestha [EMAIL PROTECTED] wrote: Hi all, i have a basic question on asterisk.The below is my scenerao. I have my sales offices around the globe.Theyare all connected with Speed Internet connection.I don't mind installing 1 asterisk box in each site.i don't mind using IP phone.i just wanted to call them for free at the cost of existing internet connectionwe have at each site.All the asterisk box will be connected with TCP/IP with one of it's NIC card having a WAN connectivity.is it possible with asterisk.Please let me know.Thanx _ Going green? See the top 12 foods to eat organic. http://green.msn.com/galleries/photos/photos.aspx?gid=164ocid=T003MSN51N 1653A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the best, Kyle bobert5064.deviantart.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] users.conf and voicemail
If you use voicemail.conf to configure the voicemail users, instead of using users.conf, you can just specify the voicemail box without an e-mail address and it will not send the e-mail notification of the voicemail to the user. I am not sure if this was your exact question, or if you wanted to send the user an e-mail, just without an extension. As far as deletion, if you are using the standard method of just attaching the message to an e-mail, then deleting the e-mail will not have any effect on the voicemail message on the asterisk server. However, you can use IMAP to send the e-mail to the user in which case it can be deleted from Asterisk directly from the e-mail client. I do not believe that you can turn this feature on or off by user, but I could be wrong. Hope this answers some of your question, let me know if I need to clarify anything. On Thu, Apr 17, 2008 at 1:14 PM, Jeremy Mann [EMAIL PROTECTED] wrote: Is there a way to specify per user attachment options for voicemail, from within users.conf? I know I can enable or disable it globally in voicemail.conf, but I have certain users that like the attachment feature, and others that don't. Also, can you enable/disable per user the deletion if it's attached? Thanks. -- This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the best, Kyle bobert5064.deviantart.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
We've been using D-Link DES-3028P and DES-3052P switches. They can supply full power to EACH port unlike the Linksys switches we've tried. They're also rock solid from our experience. -Jon - Original Message - From: Hilary Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 21, 2008 8:21:12 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Switch recommendation? On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis [EMAIL PROTECTED] wrote: The Cisco 3524 switch doesn't support 802.3af which is what your Linksys phones are going to want. Thank you for sharing Sean! When I saw them I felt a disturbance in the force, and now I know why! -- Just Hil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forking feature
As the previous person said, Asterisk will only accept one phone for each SIP account. In order to do what you are trying to do, you will want to create 2 entries in sip.conf such as [1000] and [1001] After that, you will need to set it up in extensions.conf you will create an extension that calls both SIP channels at once as shown belop exten = 1234,1,Answer() exten = 1234,n,Dial(SIP/1000SIP/1001) exten = 1234,n,Hangup() This will cause both phones to ring simultaneously. The first phone that picks up will be connected to the call, and the other will stop wringing. Hope this helps. P.S. You could even have it dump out to a common voicemail box be inserting the voicemail() application before the Hangup() line. On Thu, Apr 17, 2008 at 7:21 AM, Grey Man [EMAIL PROTECTED] wrote: Asterisk only allows a single contact per SIP account so to do forking you'll need to use two SIP accounts and put them both in the Dial command. Or you could use OpenSER. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the best, Kyle bobert5064.deviantart.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
Jonathan C. Bailey wrote: We've been using D-Link DES-3028P and DES-3052P switches. They can supply full power to EACH port unlike the Linksys switches we've tried. They're also rock solid from our experience. I echo that recommendation. The Linksys switches are probably the loudest that I've used. The D-Link's have been very reliable. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] users.conf and voicemail
On Monday 21 April 2008 20:44, Kyle Gibbons wrote: On Thu, Apr 17, 2008 at 1:14 PM, Jeremy Mann [EMAIL PROTECTED] wrote: Is there a way to specify per user attachment options for voicemail, from within users.conf? I know I can enable or disable it globally in voicemail.conf, but I have certain users that like the attachment feature, and others that don't. Also, can you enable/disable per user the deletion if it's attached? If you use voicemail.conf to configure the voicemail users, instead of using users.conf, you can just specify the voicemail box without an e-mail address and it will not send the e-mail notification of the voicemail to the user. I am not sure if this was your exact question, or if you wanted to send the user an e-mail, just without an extension. As far as deletion, if you are using the standard method of just attaching the message to an e-mail, then deleting the e-mail will not have any effect on the voicemail message on the asterisk server. However, you can use IMAP to send the e-mail to the user in which case it can be deleted from Asterisk directly from the e-mail client. I do not believe that you can turn this feature on or off by user, but I could be wrong. Hope this answers some of your question, let me know if I need to clarify anything. I think what he was looking for was confirmation that you can specify the same option names in users.conf as you can in voicemail.conf, and yes, that is the case. So just use the same option names, I think in this case, attach=yes and delete=yes, and everything will work just as if it had been specified in voicemail.conf. Note that just like in voicemail.conf, you cannot disable the actual sending of the message without blanking out the email address. Oh, and just like ODBC voicemail, the implied fields in voicemail.conf have the same name in both the database and users.conf. Those are fullname, email, and pager. So it's quite similar to other configuration interfaces in terms of naming, even though it's in users.conf. The only one that is really different is vmsecret, mainly because you do not want to change your SIP authentication password when you're trying to change your voicemail password. That would really suck, otherwise. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I roll my own E911?
Assuming I only operate in one municipality (I do), and assuming I made some sort of connection to the emergency services center in this area, via SIP or a T1 or whatever, does asterisk have a way for me to send the E911 address data? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] re-invite (bypass asterisk) post call establishment
Apologies for not explaining the set up . Using AstMan API, I Originate a call to user A. User A is a conference bridge which needs pin authentication. So post 200 OK, I need to send DTMFs for that pin. After sending the pin, I Dial (using the Originate context) user B. Now user B is behind a PBX, so I need to dial the extension for user B. I send the extension digits using DTMFs again. So, if I set canreinvite=yes, as soon as I get a 183/200 OK from user B, re-Invites are sent to both participants with the other's SDP. So, my question : once the SDPs are exchanged, what will happen to the DTMFs sent by Asterisk using sendDTMF or the D option in dial. Another scenario would be to call user B first and then user A first. The same case applies over there as well. Is there any other way to tell asterisk when to do a re-Invite/control the timing of the re-Invite? Hope I am clear this time. cheerz - Ben. Steve Davies [EMAIL PROTECTED] wrote: On 21/04/2008, Benjamin Jacob wrote: Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. You don't say what you've tried already, but as long as canreinvite=yes is set against the SIP peer, the RTP stream should be redirected once the connection is open. As far as DTMF to dial an extension at the remote end, have you looked at the D() parameter to the Dial command? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
