[asterisk-users] feedback mechanism
Hello guys, Is Asterisk capable of sending feedback to a load balancer, such as, notifying LB when maximum capacity of Asterisk server has change (like a GW with more or less E1 cards)? -- Lito A. Lampitoc -- _ -- 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
[asterisk-users] Return variables from func_odbc calls?
This is primarily aimed at Sir Lesher, whose name graces the source code for func_odbc that I'm currently trying to read to answer this question. Tilghman (or anyone else who has determined the answer to this query), I have googled, searched wikis, and I'm currently perusing the source code, but the long and short of it is that I cannot seem to find any reference to variables set by func_odbc calls such as something that would indicate if a query worked so that I can (in the dialplan) handle errors on the fly. Another item I'm trying to determine is the LAST_INSERT_ID... Thoughts/Comments? I hope very much that I haven't overlooked something, but then again I'm no longer a spring chicken either -- _ -- 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] Return variables from func_odbc calls?
On Wednesday 26 January 2011 03:02:19 Sherwood McGowan wrote: This is primarily aimed at Sir Lesher, whose name graces the source code for func_odbc that I'm currently trying to read to answer this question. Tilghman (or anyone else who has determined the answer to this query), I have googled, searched wikis, and I'm currently perusing the source code, but the long and short of it is that I cannot seem to find any reference to variables set by func_odbc calls such as something that would indicate if a query worked so that I can (in the dialplan) handle errors on the fly. Another item I'm trying to determine is the LAST_INSERT_ID... Thoughts/Comments? I hope very much that I haven't overlooked something, but then again I'm no longer a spring chicken either Well, it depends upon what type of query you're performing. If it is a query which inserts/updates, then ODBC_ROWS will contain an integer specifying the number of rows affected. -1 is reserved for a statement which failed, since it is perfectly possible for an UPDATE to succeed, yet affect 0 rows. For SELECT queries, however, that is a much more difficult question, since it depends upon the particular query. Again, it is perfectly possible for a SELECT query to successfully run, yet return 0 rows. Or it might be that with your dataset, you should never get 0 rows returned. These are questions that must be pondered by the particular data administrator, not answers that I can provide as the author of the tool. As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. -- Tilghman -- _ -- 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] Return variables from func_odbc calls?
On Wed, Jan 26, 2011 at 3:56 AM, Tilghman Lesher tilgh...@meg.abyt.es wrote: Well, it depends upon what type of query you're performing. If it is a query which inserts/updates, then ODBC_ROWS will contain an integer specifying the number of rows affected. -1 is reserved for a statement which failed, since it is perfectly possible for an UPDATE to succeed, yet affect 0 rows. For SELECT queries, however, that is a much more difficult question, since it depends upon the particular query. Again, it is perfectly possible for a SELECT query to successfully run, yet return 0 rows. Or it might be that with your dataset, you should never get 0 rows returned. These are questions that must be pondered by the particular data administrator, not answers that I can provide as the author of the tool. As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. -- Tilghman Thanks mate, that's what I was looking for :D Forgot about LAST_INSERT_ID being a MySQL-ism, but that's no big deal, I'll get by without it ;-) The ODBC_ROWS variable was what I was looking for, thanks! -- _ -- 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] Return variables from func_odbc calls?
On Wed, Jan 26, 2011 at 5:17 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Wed, Jan 26, 2011 at 3:56 AM, Tilghman Lesher tilgh...@meg.abyt.es wrote:
Well, it depends upon what type of query you're performing. If it is
a query which inserts/updates, then ODBC_ROWS will contain an
integer specifying the number of rows affected. -1 is reserved for
a statement which failed, since it is perfectly possible for an UPDATE
to succeed, yet affect 0 rows. For SELECT queries, however, that is a
much more difficult question, since it depends upon the particular query.
Again, it is perfectly possible for a SELECT query to successfully run, yet
return 0 rows. Or it might be that with your dataset, you should never get
0 rows returned. These are questions that must be pondered by the
particular data administrator, not answers that I can provide as the author
of the tool.
As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported,
since it is not portable across database types.
--
Tilghman
Thanks mate, that's what I was looking for :D Forgot about
LAST_INSERT_ID being a MySQL-ism, but that's no big deal, I'll get by
without it ;-) The ODBC_ROWS variable was what I was looking for,
thanks!
Would ${ODBCSTATUS} properly return SUCCESS or FAILED/FAILURE per
insert or update query status?
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[asterisk-users] Variable HANGUPCAUSE always empty with DAHDI
Hi,
I am using
Asterisk: 1.6.1.20
LibPRI: 1.4.11.4
DAHDI: 2.3.0.1 Echo Canceller: MG2
Wanpipe-Driver: 3.5.15
Sangoma-Firmware: 43 (A104d)
I handle some calls with my own PHP-AGI-Script. After a dial-command I
use GET FULL VARIABLE ${answeredtime} or GET FULL VARIABLE
${dialstatus} and get valid information. Sometimes dialstatus has the
value CONGESTION OR CANCEL. In this cases I tried to get the
hangupcase with GET FULL VARIABLE ${hangupcause} but I always get an
empty result.
Examples:
[...]
[Jan 25 09:23:18] VERBOSE[28228] res_agi.c: DAHDI/64-1AGI Rx GET
FULL VARIABLE ${answeredtime}
[Jan 25 09:23:18] VERBOSE[28228] res_agi.c: DAHDI/64-1AGI Tx 200
result=1 ()
[...]
[Jan 25 09:51:00] VERBOSE[1349] res_agi.c: DAHDI/33-1AGI Rx GET
FULL VARIABLE ${dialstatus}
[Jan 25 09:51:00] VERBOSE[1349] res_agi.c: DAHDI/33-1AGI Tx 200
result=1 (CANCEL)
[...]
[Jan 25 09:51:00] VERBOSE[1349] res_agi.c: DAHDI/33-1AGI Rx GET
FULL VARIABLE ${hangupcause}
[Jan 25 09:51:00] VERBOSE[1349] res_agi.c: DAHDI/33-1AGI Tx 200
result=1 ()
[...]
I expect the ISDN-Cause-Code here:
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
Does anyone have experience with that? Any idea why I do not get a valid
hangupcause?
Thanks so far,
-Thorsten-
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Re: [asterisk-users] Return variables from func_odbc calls?
On 11-01-26 04:56 AM, Tilghman Lesher wrote: As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with func_ODBC. Of cource, my database is MySQL and this function would not work on anything else. [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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] Return variables from func_odbc calls?
On Wed, Jan 26, 2011 at 7:01 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-26 04:56 AM, Tilghman Lesher wrote: As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with func_ODBC. Of cource, my database is MySQL and this function would not work on anything else. [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org Hey, thanks for the tip Paul! -- _ -- 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] Return variables from func_odbc calls?
On 11-01-26 08:19 AM, Sherwood McGowan wrote: While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with func_ODBC. Of cource, my database is MySQL and this function would not work on anything else. [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); Hey, thanks for the tip Paul! I should also note, make sure you create a 2nd DSN for your specific ODBC commands that will use LAST_INSERT_ID(), otherwise if you are using ODBC CDR or CEL, there is a chance LAST_INSERT_ID() will return the ID of those records. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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] Help determining SpanDSP version
On 01/25/2011 3:38 PM, Danny Nicholas wrote: [snip] Is there a good way to determine what version of SpanDSP I have installed and whether the app_fax.so module is the same version? [snip] Try these two commands: - whereis spandsp.so - find /|grep spandsp.so Those commands do point towards related pieces, and I think that /usr/include/spandsp/version.h might hold some clues, it doesn't shed any light on the app_fax.so module. Please pardon my ignorance in this area, I'm sure it's straightforward. As for compiling, I have started with a packaged version, and will move to rolling my own as things move along. Many thanks, Tom -- _ -- 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] feedback mechanism
On 11-01-26 03:36 AM, Lito Lampitoc wrote: Hello guys, Is Asterisk capable of sending feedback to a load balancer, such as, notifying LB when maximum capacity of Asterisk server has change (like a GW with more or less E1 cards)? Within Asterisk, no. However you could write a script that polls the AMI for information, then sends it to the load balancer. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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
[asterisk-users] Recommended Windows client to display CID?
Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. -- _ -- 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] Recommended Windows client to display CID?
On 01/26/2011 08:52 AM, Gilles wrote: If you like open source what are you doing running windows ? Getting anything to work properly there which does network communications is a huge PITA since every user has their own firewall and different settings etc etc etc. Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. -- _ -- 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] Recommended Windows client to display CID?
I have asterisk call out to a shell script which sends a jabber message to the user (along with links to any open tickets in our ticketing system associated with that CID). All free, but requires work to build. On Jan 26, 2011, at 6:52 AM, Gilles codecompl...@free.fr wrote: Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. -- _ -- 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] Recommended Windows client to display CID?
On Wed, 26 Jan 2011 09:04:30 -0500, jon pounder j...@inline.net wrote: If you like open source what are you doing running windows ? Right, but it's just that NetCID is supposed to be used as an add-on to its commercial Identify application, so it's illegal to redistribute. Getting anything to work properly there which does network communications is a huge PITA since every user has their own firewall and different settings etc etc etc. Provided the users' all live in the same LAN and have their firewall configured in such as way as to either receive UDP datagrams or connect out to Asterisk's AMI... are there good, open-source solutions to display CID information on users' monitor? Thank you. -- _ -- 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] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt
On 11-01-24 07:28 PM, Doug wrote: Does anyone know how to get rid of these warnings? Disable NOTICE within logger.conf? They are just information about the status of SIP Subscriptions. Post an example log of showing the frequency, it maybe possible to change them to DEBUG if they are too noisy. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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] Recommended Windows client to display CID?
On Wednesday 26 Jan 2011, Gilles wrote: I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. A web browser on the Windows boxes; continually refreshing a CGI script hosted on the Asterisk server that displays data from a database, which in turn is updated by an AGI script run from within the dialplan whenever a call comes in or hangs up? Otherwise, you'd need to run some sort of daemon on the Windows boxes that listened out for messages sent by an AGI script. Maybe you can hack an instant messaging client to pieces for this purpose? -- 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] Recommended Windows client to display CID?
On Wed, 26 Jan 2011 14:17:37 +, A J Stiles asterisk_l...@earthshod.co.uk wrote: A web browser on the Windows boxes; continually refreshing a CGI script hosted on the Asterisk server that displays data from a database, which in turn is updated by an AGI script run from within the dialplan whenever a call comes in or hangs up? Thanks for the idea, but I'd rather display a pop-up than updating a web page, because the notification is more obvious to the user. Otherwise, you'd need to run some sort of daemon on the Windows boxes that listened out for messages sent by an AGI script. Maybe you can hack an instant messaging client to pieces for this purpose? Before hacking my own, I'd like to make sure there isn't a good, available solution. I'll check if there's an open-source, light, easy-to-deploy Jabber client that I could use for this purpose. Thank you. -- _ -- 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
[asterisk-users] Caching CALLERID(dnid)
Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read the variable CALLERID(dnid) with AMI. This call will be ended. - Then we make an outbound call on the same channel. The CALLERID(dnid) is not set, during this outbound call. If this outbound call is picked up, we will read the CALLERID(dnid) with AMI. Now we see that the CALLERID(dnid) is still '655871460' Is there a way to reset the CALLERID(dnid) on one channel or automatically reset the complete cache on one channel if this channel is ended? Regards, Ami command: action: GetVar actionid: 129675971_656137# variable: CALLERID(dnid) channel: DAHDI/11-1 Arjan Kroon Mobillion BV -- _ -- 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] ReceiveFAX issue.
On Tue, Jan 25, 2011 at 7:01 PM, Bryant Zimmerman brya...@zktech.com wrote: Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't make t.38 work I keep getting the following error Disconnected after permitted retries Any ideas on this? So you're saying if you turn off t38 in sip.conf, you receive faxes successfully? Problem solved. Don't use T.38 in your particular environment. -- _ -- 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] Return variables from func_odbc calls?
I actually was pondering that same thing :D On Wed, Jan 26, 2011 at 7:33 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-26 08:19 AM, Sherwood McGowan wrote: While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with func_ODBC. Of cource, my database is MySQL and this function would not work on anything else. [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); Hey, thanks for the tip Paul! I should also note, make sure you create a 2nd DSN for your specific ODBC commands that will use LAST_INSERT_ID(), otherwise if you are using ODBC CDR or CEL, there is a chance LAST_INSERT_ID() will return the ID of those records. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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] Recommended Windows client to display CID?
On 01/26/2011 9:04 AM, jon pounder wrote: On 01/26/2011 08:52 AM, Gilles wrote: If you like open source what are you doing running windows ? Getting anything to work properly there which does network communications is a huge PITA since every user has their own firewall and different settings etc etc etc. Unless, of course, you properly implement Group Policies (which is Windows Server only, IIRC, but still...) On 01/26/2011 9:14 AM, Joel Maslak wrote: I have asterisk call out to a shell script which sends a jabber message to the user (along with links to any open tickets in our ticketing system associated with that CID). All free, but requires work to build. Ooh. I like this. Can you post a sample, or maybe a synopsis of what pieces you are using to tie this all together? To answer the OP's question about XMPP clients, Spark from Ignite Realtime and Pandion are both good in my experience. Tom -- _ -- 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] ReceiveFAX issue.
David Backeberg dbackeb...@gmail.com writes: So you're saying if you turn off t38 in sip.conf, you receive faxes successfully? Problem solved. Don't use T.38 in your particular environment. That is not particularly useful advice. Fax over VoIP without T.38 is inherently unreliable except in very controlled environments. That a few faxes happen to work does not make T.38 a bad choice. It is in the interest of the Asterisk community to fix whichever bug/incompatibility Bryant Zimmerman is hitting. Of course in reality no one with the right skills may have the time to do so. Again, that does not make the problem solved. /Benny -- _ -- 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] Recommended Windows client to display CID?
On Wed, 26 Jan 2011 09:55:23 -0500, Tom Rymes try...@rymes.com wrote: Unless, of course, you properly implement Group Policies (which is Windows Server only, IIRC, but still...) Are there other issues to expect besides having to configure Windows' firewall to allow either UDP broadcasts or outgoing TCP connections to AMI? To answer the OP's question about XMPP clients, Spark from Ignite Realtime and Pandion are both good in my experience. Thanks, I'll check 'em out. -- _ -- 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
[asterisk-users] Pickup local/.... not working
Hello list, is it possible that it is not possible to pickup a local channel ?? [Jan 26 16:13:43] -- Executing [10@sub-pickup:24] Pickup(SIP/voip5-0750, Local/329596@default-505a;2@PICKUPMARK) in new stack [Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec: No target channel found for Local/329596. This is an incoming call that rings extension 10, and I want to pickup this call from another extension. It goes well when the call is coming from external. This call however never leaves Asterisk so Asterisk creates a Local/... channel. But pickup fails here... Kind regards, Jonas. -- _ -- 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] ReceiveFAX issue.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Wednesday, January 26, 2011 9:12 AM To: David Backeberg Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ReceiveFAX issue. David Backeberg dbackeb...@gmail.com writes: So you're saying if you turn off t38 in sip.conf, you receive faxes successfully? Problem solved. Don't use T.38 in your particular environment. That is not particularly useful advice. Fax over VoIP without T.38 is inherently unreliable except in very controlled environments. That a few faxes happen to work does not make T.38 a bad choice. It is in the interest of the Asterisk community to fix whichever bug/incompatibility Bryant Zimmerman is hitting. Of course in reality no one with the right skills may have the time to do so. Again, that does not make the problem solved. /Benny Maybe it's just me, but it seems that Fax over VOIP is almost a step back and that T.38 is an excruciatingly painful venture down that path. The skills here run the entire gamut from just over pushing buttons on a calculator to Space Shuttle Engineers. The problem solved answer is somewhat flippant, but T.38 Faxing is a small but vocal segment of the posters here. It would be nice to have a universally proven Faxing solution for Asterisk in all flavors/technologies, but I'm not holding my breath on that one. -- _ -- 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] Pickup local/.... not working
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 26, 2011 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pickup local/ not working Hello list, is it possible that it is not possible to pickup a local channel ?? [Jan 26 16:13:43] -- Executing [10@sub-pickup:24] Pickup(SIP/voip5-0750, Local/329596@default-505a;2@PICKUPMARK) in new stack [Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec: No target channel found for Local/329596. This is an incoming call that rings extension 10, and I want to pickup this call from another extension. It goes well when the call is coming from external. This call however never leaves Asterisk so Asterisk creates a Local/... channel. But pickup fails here... Kind regards, Jonas. As I understand it, Local channels are work channels for asterisk to hold calls or perform tasks in and are therefore not pickable. You would have to transfer or bridge the channel to a real (SIP/DAHDI) channel to actually use it. -- _ -- 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] ReceiveFAX issue.
Has anyone else seen an issue with t.38 faxing on Level 3 with res_fax and res_fax_spandsp.so What we are seeing in the packet captuers is that the call is trying to do t.38 but does not appear to be completing the handshaking. No data is being transmitted. I have included a link to my pcap of this. Can anyone give me some more insight? cap-t38.pcap Thanks Bryant -- _ -- 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
[asterisk-users] Regarding error in Asterisk dail plan:
Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i am getting error like this NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default' i dont know y this is coming its really troubling me a lot... please any one send me some xml, dial plan and sip.conf files for registering and for inviting. I have been trying for this a lot if any one help me i would be more thankful . BR viswavardhanreddy -- _ -- 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] Pickup local/.... not working
On 01/26/2011 04:26 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 26, 2011 9:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Pickup local/ not working Hello list, is it possible that it is not possible to pickup a local channel ?? [Jan 26 16:13:43] -- Executing [10@sub-pickup:24] Pickup(SIP/voip5-0750, Local/329596@default-505a;2@PICKUPMARK) in new stack [Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec: No target channel found for Local/329596. This is an incoming call that rings extension 10, and I want to pickup this call from another extension. It goes well when the call is coming from external. This call however never leaves Asterisk so Asterisk creates a Local/... channel. But pickup fails here... Kind regards, Jonas. As I understand it, Local channels are work channels for asterisk to hold calls or perform tasks in and are therefore not pickable. You would have to transfer or bridge the channel to a real (SIP/DAHDI) channel to actually use it. I have something like : exten = 329596,1,GoTo(newcontext,329596,1) [newcontext] exten = 10,1,Dial(SIP/MySipAccount) This makes a call to 329596 go to the context newcontext, but Asterisk creates a Local/... channel for this re-direct to another context. Is there then a way to transfer the call to another context ?? Kind regards, Jonas. -- _ -- 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] Pickup local/.... not working
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 26, 2011 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pickup local/ not working On 01/26/2011 04:26 PM, Danny Nicholas wrote: _ size=2 width=100% align=center From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 26, 2011 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pickup local/ not working Hello list, is it possible that it is not possible to pickup a local channel ?? [Jan 26 16:13:43] -- Executing [10@sub-pickup:24] Pickup(SIP/voip5-0750, Local/329596@default-505a;2@PICKUPMARK) in new stack [Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec: No target channel found for Local/329596. This is an incoming call that rings extension 10, and I want to pickup this call from another extension. It goes well when the call is coming from external. This call however never leaves Asterisk so Asterisk creates a Local/... channel. But pickup fails here... Kind regards, Jonas. As I understand it, Local channels are work channels for asterisk to hold calls or perform tasks in and are therefore not pickable. You would have to transfer or bridge the channel to a real (SIP/DAHDI) channel to actually use it. I have something like : exten = 329596,1,GoTo(newcontext,329596,1) [newcontext] exten = 10,1,Dial(SIP/MySipAccount) This makes a call to 329596 go to the context newcontext, but Asterisk creates a Local/... channel for this re-direct to another context. Is there then a way to transfer the call to another context ?? Kind regards, Jonas. I would have done it this way exten = 329596,1,GoTo(newcontext,s,1) [newcontext] exten = s,1,Dial(SIP/MySipAccount) This should open a SIP channel when you dial 329596. Hope this helps. -- _ -- 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] Regarding error in Asterisk dail plan:
On 11-01-26 11:28 AM, viswavardhanreddy karna wrote: Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... Asterisk is not a SIP Proxy, it is a B2BUA. NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default' You can troubleshoot dialplan issues pretty easy with: *CLI dialplan show service@default The message states, exten = service,1,blah() is missing from the [default] context. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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] Regarding error in Asterisk dail plan:
From: viswavardhanreddy karna viswavardhanre...@gmail.com Sent: Wednesday, January 26, 2011 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Regarding error in Asterisk dail plan: Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_usi ng_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i am getting error like this NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default' i dont know y this is coming its really troubling me a lot... please any one send me some xml, dial plan and sip.conf files for registering and for inviting. I have been trying for this a lot if any one help me i would be more thankful . BR viswavardhanreddy - viswavardhanreddy Your inbound request is not being sent with any target context or it is not matching the ip found in your sip peers. This causes the default context to trying and handle the call and you don't have anthing in it that can complete the call. The three options are 1 if you are doing registration make sure that the sending device is specifiying a context. (It does not look like you are based on your link) 2 make sure that the sending ip matches your peer account or change the peer account to friend (also change your peers to use insecure=port,invite and see if that helps) 3 add a universal handler to the [default] contect to direct the call to your test contects (exten = _.X,1,Goto(test,s,1) One of these ideas may help you if I am understanding your issue. Bryant -- _ -- 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] Pickup local/.... not working
I think you're missing something in your explanation... the code represented in your email shows no reason for a Local channel to be recreated. Goto commands do not result in Local channel creation, nor does the Dial command On Wed, Jan 26, 2011 at 10:34 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 01/26/2011 04:26 PM, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 26, 2011 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pickup local/ not working Hello list, is it possible that it is not possible to pickup a local channel ?? [Jan 26 16:13:43] -- Executing [10@sub-pickup:24] Pickup(SIP/voip5-0750, Local/329596@default-505a;2@PICKUPMARK) in new stack [Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec: No target channel found for Local/329596. This is an incoming call that rings extension 10, and I want to pickup this call from another extension. It goes well when the call is coming from external. This call however never leaves Asterisk so Asterisk creates a Local/... channel. But pickup fails here... Kind regards, Jonas. As I understand it, Local channels are “work” channels for asterisk to hold calls or perform tasks in and are therefore not “pickable”. You would have to transfer or bridge the channel to a “real” (SIP/DAHDI) channel to actually use it. I have something like : exten = 329596,1,GoTo(newcontext,329596,1) [newcontext] exten = 10,1,Dial(SIP/MySipAccount) This makes a call to 329596 go to the context newcontext, but Asterisk creates a Local/... channel for this re-direct to another context. Is there then a way to transfer the call to another context ?? Kind regards, Jonas. -- _ -- 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] Recommended Windows client to display CID?
On 11-01-26 08:52 AM, Gilles wrote: Hello I'd like to display CID information on users' monitor running Windows. You could use any XMPP client and send a message to it using JabberSend() from the dialplan. We document using it at http://ofps.oreilly.com. Leif. -- _ -- 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] Regarding error in Asterisk dail plan:
On Wed, Jan 26, 2011 at 9:28 AM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i am getting error like this NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default' i dont know y this is coming its really troubling me a lot... viswavardhanreddy-- I'm sorry, I'm a bit tight on time, I haven't read your link. But I did some performance testing of Asterisk some years ago, and wrote a doc about it and it's part of the source tree of Asterisk (At least in 1.6 ). See doc/chan_sip-perf-testing.txt There I show how I tied sipp and asterisk together. It might not at all help you, might not be your approach at all, but it might give you some ideas. Best of luck! murf -- Steve Murphy -- _ -- 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] Regarding error in Asterisk dail plan:
thanks to all, but i am working for register scenario can anyone please help me when i have sent the sipp command from sipp like this ./sipp -sf reg.xml -inf users.csv -p 5060 -i 192.168.1.99 192.168.1.100 i got the error message in asterisk like this chan_sip.c:21819 handle_request_register: Registration from '105 sip:105@192.168.1.100:5060' failed for '192.168.1.99' - No matching peer found and in wire shark i got 404 not found. anyone please help me. On Wed, Jan 26, 2011 at 6:35 PM, Steve Murphy m...@parsetree.com wrote: On Wed, Jan 26, 2011 at 9:28 AM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i am getting error like this NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default' i dont know y this is coming its really troubling me a lot... viswavardhanreddy-- I'm sorry, I'm a bit tight on time, I haven't read your link. But I did some performance testing of Asterisk some years ago, and wrote a doc about it and it's part of the source tree of Asterisk (At least in 1.6 ). See doc/chan_sip-perf-testing.txt There I show how I tied sipp and asterisk together. It might not at all help you, might not be your approach at all, but it might give you some ideas. Best of luck! murf -- Steve Murphy -- _ -- 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] Regarding error in Asterisk dail plan:
On Wed, 26 Jan 2011, Bryant Zimmerman wrote:
3 add a universal handler to the [default] contect to direct the call to your
test contects (exten =
_.X,1,Goto(test,s,1)
exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n, goto(test,s,1)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
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Re: [asterisk-users] Recommended Windows client to display CID?
On Wed, 26 Jan 2011 12:15:01 -0500, Leif Madsen leif.mad...@asteriskdocs.org wrote: You could use any XMPP client and send a message to it using JabberSend() from the dialplan. We document using it at http://ofps.oreilly.com. Thanks Leif. -- _ -- 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
[asterisk-users] Really wacky problem with internal extensions.
We have an Asterisk server acting as a hosted PBX system for many clients,
and we're going through an upgrade to Asterisk 1.6 by moving our most
important (and complicated) clients one at a time.
But we're having a problem with one customer that I really can't explain.
I can place calls directly to one phone at the customer's location (they
also have an IVR that asks for an extension number), but the customer
cannot do the same internally. All other outbound calls from this
customer, work. The dialplans for the IVR and for internal dialing are
very nearly identical, and making them completely identical doesn't change
anything. The dialplans are pasted at the end of this message.
When the customer dials an internal extension, the Asterisk console
produces this output (usernames redacted):
-- Executing [303@XX:1] Set(SIP/XX2-04ce,
CALLERID(name)=Internal call) in new stack
-- Executing [303@XX:2] GotoIf(SIP/XX2-04ce,
0?dialfw:dial) in new stack
-- Goto (XX,303,8)
-- Executing [303@XX:8] Dial(SIP/XX2-04ce,
SIP/XX3,20,g) in new stack
== Using SIP RTP CoS mark 5
-- Called XX3
-- Got SIP response 400 Bad Request back from 209.53.201.33
-- SIP/XX3-04cf is circuit-busy
Usually this SIP response 400 error is due to the firewall at the
customer's location blocking the incoming connection, but then why would
normal inbound calls work? It's not like the Dial() command for those
inbound calls is any different. This customer hasn't changed any firewall
rules during the changeover, and is forwarding unique ports for each
phone. Furthermore, the SIP configuration for these phones send a
qualification message every 60 seconds to keep any NAT translation alive.
Anyway, here's the dialplan for the IVR (only extensions 302 and 303 are
included for brevity):
[ivr-XX]
exten = s,1,Answer
exten = s,n,Playback(silence/1)
exten = s,n,Background(XX/greeting)
exten = s,n,WaitExten(4)
exten = 302,1,GotoIf(${DB_EXISTS(CFIM/302)}?dialfw:dial)
exten = 302,n(dialfw),Set(extension=${DB(CFIM/302)})
exten = 302,n,Set(wait=${MATH(${DB(NumRing/302)}*6,int)})
exten = 302,n,ExecIf($[${wait} != 0]|Dial|SIP/XX2|${wait}|g|)
exten = 302,n,Dial(DAHDI/g1/${extension},90,g)
exten = 302,n,Macro(handle-hangup)
exten = 302,n(dial),Dial(SIP/XX2,30,g)
exten = 302,n,Voicemail(302,u)
exten = 302,n,Macro(handle-hangup)
exten = 303,1,GotoIf(${DB_EXISTS(CFIM/303)}?dialfw:dial)
exten = 303,n(dialfw),Set(extension=${DB(CFIM/303)})
exten = 303,n,Set(wait=${MATH(${DB(NumRing/303)}*6,int)})
exten = 303,n,ExecIf($[${wait} != 0]|Dial|SIP/XX3|${wait}|g|)
exten = 303,n,Dial(DAHDI/g1/${extension},90,g)
exten = 303,n,Macro(handle-hangup)
exten = 303,n(dial),Dial(SIP/XX3,30,g)
exten = 303,n,Voicemail(303,u)
exten = 303,n,Macro(handle-hangup)
exten = 0,1,Answer
exten = 0,n,SIPAddHeader(Alert-Info: info=Bellcore-dr4)
exten =
0,n,Dial(SIP/XX2SIP/XX3SIP/XX4SIP/XX5SIP/XX6SIP/XX7,25,g)
exten = 0,n,Voicemail(300,u)
exten = 0,n,Macro(handle-hangup)
exten = t,1,Answer
exten = t,n,SIPAddHeader(Alert-Info: info=Bellcore-dr4)
exten =
t,n,Dial(SIP/XX2SIP/XX3SIP/XX4SIP/XX5SIP/XX6SIP/XX7,25,g)
exten = t,n,Voicemail(300,u)
exten = t,n,Macro(handle-hangup)
exten = i,1,Playback(XX/invalid)
exten = i,n,Goto(s,1)
And this is the outgoing dialplan for the customer (for internal lines and
special features)
[XX]
exten = _*98,1,Answer
exten = _*98,n,VoicemailMain()
exten = _*88,1,Answer
exten = _*88,n,VoicemailMain(300)
exten = _*72,1,Answer
exten = _*72,n,Wait(1)
exten = _*72,n,Read(extension,XX/enter-extension,3)
exten = _*72,n,Read(fwdnum,XX/forward-to,10)
exten = _*72,n,Read(numrings,XX/num-of-rings,1)
exten = _*72,n,Set(DB(CFIM/${extension})=${fwdnum})
exten = _*72,n,NoOp(Numrings: ${numrings} ${numrings})
exten = _*72,n,Set(DB(NumRing/${extension})=${numrings})
exten = _*72,n,Playback(XX/your-extension)
exten = _*72,n,SayDigits(${extension})
exten = _*72,n,Playback(XX/will-forward-to)
exten = _*72,n,SayDigits(${fwdnum})
exten = _*72,n,Playback(XX/after)
exten = _*72,n,SayDigits(${numrings})
exten = _*72,n,Playback(XX/rings)
exten = _*72,n,Macro(handle-hangup)
exten = _*73,1,Answer
exten = _*73,n,Wait(1)
exten = _*73,n,Read(extension,XX/enter-extension,3)
exten = _*73,n,Set(${DB_DELETE(CFIM/${extension}))
exten = _*73,n,Playback(XX/cfwd-cancelled)
exten = _*73,n,Macro(handle-hangup)
exten = 302,1,Set(CALLERID(name)=Internal call)
exten = 302,n,GotoIf(${DB_EXISTS(CFIM/302)}?dialfw:dial)
exten = 302,n(dialfw),Set(extension=${DB(CFIM/302)})
exten = 302,n,Set(wait=${MATH(${DB(NumRing/302)}*6,int)})
exten = 302,n,ExecIf($[${wait} != 0]|Dial,SIP/XX2,${wait},g)
exten = 302,n,Dial(DAHDI/g1/${extension},90,g)
exten = 302,n,Macro(handle-hangup)
exten = 302,n(dial),Dial(SIP/XX2,20,g)
exten = 302,n,Voicemail(302,u)
exten = 302,n,Macro(handle-hangup)
exten =
Re: [asterisk-users] Regarding error in Asterisk dail plan:
Hi edwards,
i have taken register.xml and csv file from this site
http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_beta
http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_betai
have written sip.conf and extension.conf i got error could you plz help me
by writing the sip.conf and extensions.conf
plz send me some file regarding this
On Wed, Jan 26, 2011 at 6:53 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 26 Jan 2011, Bryant Zimmerman wrote:
3 add a universal handler to the [default] contect to direct the call
to your test contects (exten =
_.X,1,Goto(test,s,1)
exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n, goto(test,s,1)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Recommended Windows client to display CID?
On Wed, Jan 26, 2011 at 1:28 PM, Asterisk l...@halfmind.com wrote: On 01/26/2011 9:14 AM, Joel Maslak wrote: I have asterisk call out to a shell script which sends a jabber message to the user (along with links to any open tickets in our ticketing system associated with that CID). All free, but requires work to build. Ooh. I like this. Can you post a sample, or maybe a synopsis of what pieces you are using to tie this all together? This can also be accomplished using JabberSend. -- _ -- 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] res_fax
Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] res_fax
On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] ReceiveFAX issue.
On 01/26/2011 10:27 AM, Bryant Zimmerman wrote: Has anyone else seen an issue with t.38 faxing on Level 3 with res_fax and res_fax_spandsp.so What we are seeing in the packet captuers is that the call is trying to do t.38 but does not appear to be completing the handshaking. No data is being transmitted. I have included a link to my pcap of this. Can anyone give me some more insight? cap-t38.pcap http://webmail.zktech.com/public/downloadfile.aspx?f=ulHIhepag5qoKm0cTUmljmT%2f7YCcOPvzlyZcnZg%2fG2B25W%2fsSr6Uwbu%2bET3kbKw84pTJjtuqrPQ%3d res_fax_spandsp (and res_fax_digium) are not involved with T.38 negotiation; that is handled by res_fax and chan_sip. As I've posted in the other thread you started, please post a complete Asterisk log capture so that we can see what Asterisk tried to do and what the results were. That is more useful than a packet capture, at least for the initial debugging steps. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] Regarding error in Asterisk dail plan:
Un-top-posting... On Wed, 26 Jan 2011, viswavardhanreddy karna wrote: Hi all, i am doing my master thesis on server perfromance On Wed, 26 Jan 2011, viswavardhanreddy karna wrote: i have taken register.xml and csv file from this site http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_beta i have written sip.conf and extension.conf i got error could you plz help me by writing the sip.conf and extensions.conf Something about 'doing my master thesis' makes me think this is something you should solve with a bit more effort on your part. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] res_fax
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Steve did not write res_fax (which where SendFAX and ReceiveFAX come from) snip I am personally a little confused here, because I have a ReceiveFAX application when I unload the res_fax module and res_fax_digium module and load the app_fax module. In other words, I think that multiple modules provide applications named ReceiveFax and SendFAX. Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? Tom -- _ -- 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] res_fax
On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Steve did not write res_fax (which where SendFAX and ReceiveFAX come from) snip I am personally a little confused here, because I have a ReceiveFAX application when I unload the res_fax module and res_fax_digium module and load the app_fax module. In other words, I think that multiple modules provide applications named ReceiveFax and SendFAX. Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? That is correct. app_fax is deprecated (and that is why it is marked as don't build by default), and res_fax plus a technology module (res_fax_spandsp or res_fax_digium) is the replacement for it. All of the work that the Digium team has done improving T.38 negotiation and interoperability has gone into res_fax, not app_fax. Users of Asterisk 1.8.x should only choose to build app_fax if they have a specific need for it (and if that's the case we'd like to know what the need is so we can ensure that res_fax can satisfy it). Users of older Asterisk releases will have app_fax by default (since res_fax was not included in those versions), but if they want to use Digium's res_fax_digium module they can download it along with res_fax and use them instead. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 1:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt cap-t38.pcap And by the way thank you for your response it is appreciated. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- 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] Help determining SpanDSP version
Hello TOm ( all) , ldd -v app_fax.so Should list all items linked against in the module . Hth , JimL On Wed, 26 Jan 2011, Tom Rymes wrote: On 01/25/2011 3:38 PM, Danny Nicholas wrote: [snip] Is there a good way to determine what version of SpanDSP I have installed and whether the app_fax.so module is the same version? [snip] Try these two commands: - whereis spandsp.so - find /|grep spandsp.so Those commands do point towards related pieces, and I think that /usr/include/spandsp/version.h might hold some clues, it doesn't shed any light on the app_fax.so module. Please pardon my ignorance in this area, I'm sure it's straightforward. As for compiling, I have started with a packaged version, and will move to rolling my own as things move along. Many thanks, Tom -- _ -- 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 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- 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] res_fax
On 01/26/2011 2:16 PM, Kevin P. Fleming wrote: On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? That is correct. app_fax is deprecated (and that is why it is marked as don't build by default), and res_fax plus a technology module (res_fax_spandsp or res_fax_digium) is the replacement for it. All of the work that the Digium team has done improving T.38 negotiation and interoperability has gone into res_fax, not app_fax. Users of Asterisk 1.8.x should only choose to build app_fax if they have a specific need for it (and if that's the case we'd like to know what the need is so we can ensure that res_fax can satisfy it). Users of older Asterisk releases will have app_fax by default (since res_fax was not included in those versions), but if they want to use Digium's res_fax_digium module they can download it along with res_fax and use them instead. Gotcha. So, 1.6 users who install FFA get res_fax and res_fax_digium. Presumably, 1.6 users could also combine res_fax and res_fax_spandsp? Steve - Will compiling the latest version of SpanDSP on a 1.6 system result in a res_fax_spandsp.so module? Tom -- _ -- 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] res_fax
On 01/26/2011 01:21 PM, Tom Rymes wrote: On 01/26/2011 2:16 PM, Kevin P. Fleming wrote: On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? That is correct. app_fax is deprecated (and that is why it is marked as don't build by default), and res_fax plus a technology module (res_fax_spandsp or res_fax_digium) is the replacement for it. All of the work that the Digium team has done improving T.38 negotiation and interoperability has gone into res_fax, not app_fax. Users of Asterisk 1.8.x should only choose to build app_fax if they have a specific need for it (and if that's the case we'd like to know what the need is so we can ensure that res_fax can satisfy it). Users of older Asterisk releases will have app_fax by default (since res_fax was not included in those versions), but if they want to use Digium's res_fax_digium module they can download it along with res_fax and use them instead. Gotcha. So, 1.6 users who install FFA get res_fax and res_fax_digium. Presumably, 1.6 users could also combine res_fax and res_fax_spandsp? Steve - Will compiling the latest version of SpanDSP on a 1.6 system result in a res_fax_spandsp.so module? No, res_fax_spandsp is not part of SpanDSP (but it uses SpanDSP), and we don't distribute an Asterisk 1.6.x version of res_fax_spandsp.c. It wouldn't be hard for someone to make one, though, and the res_fax binary module download does include res_fax.h so it is possible to compile against it if they wanted to do so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] res_fax
On 01/26/2011 01:19 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 1:50 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt http://webmail.zktech.com/public/downloadfile.aspx?f=KERoF6PWf6e2FK8S5zgEDs02rFGdd7zE0AIG7tjbCR9a06oFY1NwFap58zDWva3BcdOp%2b%2f%2fuBo8%3d cap-t38.pcap http://webmail.zktech.com/public/downloadfile.aspx?f=ulHIhepag5qoKm0cTUmljmT%2f7YCcOPvzlyZcnZg%2fG2B25W%2fsSr6Uwbu%2bET3kbKw84pTJjtuqrPQ%3d Unfortunately that log capture is incomplete; it doesn't include any of the messages that res_fax emits as it goes through T.38 negotiations. Please ensure that your 'console' channel in logger.conf has 'debug,verbose,warning,notice,error,fax' enabled and that you have 'core set verbose 10' and 'core set debug 10' set before the call attempt begins (or at least before ReceiveFAX is executed). If the server is only processing this particular call, then 'sip set debug on' would also be helpful. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] Recommended Windows client to display CID?
On Wed, Jan 26, 2011 at 7:55 AM, Tom Rymes try...@rymes.com wrote:
Ooh. I like this. Can you post a sample, or maybe a synopsis of what pieces
you are using to tie this all together?
I have a two processes - one to notify on an internal incoming call,
one to notify on tickets (both on incoming and outgoing calls).
The notify on incoming call just does the basic CID information. I
have a dialplan line like:
exten = _XXX,1,System(/usr/local/bin/notify_incoming_cid.pl ${EXTEN}
${CALLERID(NUMBER)} ${CALLERID(NAME)} )
This is a Perl script that reads a text file listing extension # and
Jabber ID associations, and, if it finds an association, calls a
second Perl script to send a Jabber notification, using Net::Jabber.
In addition to this, any time a call is placed, a line like the
following executes:
exten = _X.,n,System(/usr/local/bin/notify_it_jira_users.pl
${CALLERID(NUMBER)} ${ext} )
This script uses a similar method to above, but only generates a
notification if a Jira (our ticketting system) user ID associated (via
another text file) with a phone number is a reporter on any open
Jira issues (it does this via a web query to our ticketting system).
If this user is a reporter, the other leg of the call (whether
incoming or outgoing) gets a IM with a link to the specific issues
along with the summary of each issue.
For instance, if someone calls the IT department, they'll get
something like this:
---
303-555-0010 John Smith
---
CALL FROM jsmith WITH TICKETS:
http://jira/IT-1010 - Cannot log into VPN
http://jira/IT-1020 - Computer making strange sounds
---
If they don't have any open tickets, they won't get the second message
listing tickets.
We can generate the text files this solution uses automatically by
looking at our phone list database and our customer database in a Cron
tab (it would be possible to query directly the database, but this was
simpler to implement in an afternoon).
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Re: [asterisk-users] Return variables from func_odbc calls?
On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: On 11-01-26 04:56 AM, Tilghman Lesher wrote: As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with func_ODBC. Of cource, my database is MySQL and this function would not work on anything else. [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); That assumes you have only one call in existence at a time. If two calls came in and executed the query at about the same time, it's possible for both reads to return the same value. -- Tilghman -- _ -- 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
[asterisk-users] Asterisk 1.8.2.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by mnicholson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3 Thank you for your continued support of Asterisk! -- _ -- 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
[asterisk-users] list of errorswhile registering client at asterisk with sipp
Hi every one, Hello i am doing project on evaluating the sip proxy performances like asterisk, openims and opensips using the traffic generator SIPp. I am using 2 computers of same configuration as SIPp clients one as uac and other as uas... and one laptop for asterisk server.. UAC:192.168.1.99Asterisk server(192.168.1.100)---UAS:192.168.1.101 Registering: UAC:192.168.1.99Asterisk server(192.168.1.100) i am getting error in SIPp as : aborting call on unexpected message for call-id '1-4541'@192.168.1.99 1-4541%27@192.168.1.99':while sending (index 3), reveived 'SIP/2.0 200 ok in asterisk i am getting error as: [Jan 26 21:30:35] NOTICE[5188]: chan_sip.c:21819 handle_request_register: Registration from '' failed for '192.168.1.99' - No matching peer found [Jan 26 21:30:35] NOTICE[5188]: chan_sip.c:13163 register_verify: Invalid to address: '' from 192.168.1.99 (missing sip:) trying to use anyway... [Jan 26 21:30:35] NOTICE[5188]: chan_sip.c:21819 handle_request_register: Registration from '' failed for '192.168.1.99' - No matching peer found [Jan 26 21:30:35] NOTICE[5188]: chan_sip.c:13163 register_verify: Invalid to address: '' from 192.168.1.99 (missing sip:) trying to use anyway... [Jan 26 21:30:35] NOTICE[5188]: chan_sip.c:21819 handle_request_register: Registration from '' failed for '192.168.1.99' - No matching peer found [Jan 26 21:30:36] NOTICE[5188]: chan_sip.c:13163 register_verify: Invalid to address: '' from 192.168.1.99 (missing sip:) trying to use anyway... [Jan 26 21:30:36] NOTICE[5188]: chan_sip.c:21819 handle_request_register: Registration from '' failed for '192.168.1.99' - No matching peer found [Jan 26 21:30:36] NOTICE[5188]: chan_sip.c:13163 register_verify: Invalid to address: '' from 192.168.1.99 (missing sip:) trying to use anyway... [Jan 26 21:30:36] NOTICE[5188]: chan_sip.c:21819 handle_request_register: Registration from '' failed for '192.168.1.99' - No matching peer found [Jan 26 21:30:36] NOTICE[5188]: chan_sip.c:13163 register_verify: Invalid to address: '' from 192.168.1.99 (missing sip:) trying to use anyway... when i have taken trace from wire shark i got error message as 404 Not found Below i am sending my sip.conf and extensions.conf files please suggest me some help ... EXTENSIONS.CONF [others] [testing] exten=bob,1,Dial(SIP/bob) ## SIP.conf [general] port = 5060 bindaddr = 0.0.0.0 context = others [bob] type=friend context=testing host=dynamic user=bob secret=test canreinvite=no disallow=all nat=yes ALSO CHECK MY BOBREG.XML FILE FOR ERRORS .. Awaiting for the reply as soon as possible Best Regards, viswavardhanredy ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd scenario name=registration send retrans=500 ![CDATA[ REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 20 From: [field0] sip:[field0]@[local_ip]:[local_port];tag=[call_number] To: [field0] sip:[field0]@[remote_ip]:[remote_port] Call-ID: [call_id] CSeq: 1 REGISTER Contact: sip:[field0]@[local_ip]:[local_port] Expires: 3600 Content-Length: 0 User-Agent: Sipp v1.1-TLS, version 20061124 ]] /send recv response=401 auth=true rtd=true /recv send retrans=500 ![CDATA[ REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 20 From: [field0] sip:[field0]@[local_ip];tag=[call_number] To: [field0] sip:[field0]@[remote_ip] Call-ID: [call_id] CSeq: 2 REGISTER Contact: sip:[field0]@[local_ip]:[local_port] Expires: 3600 Content-Length: 0 User-Agent: Sipp v1.1-TLS, version 20061124 [field1] ]] /send send ![CDATA[ SIP/2.0 200 OK Via: SIP/2.0/[transport] [remote_ip]:[remote_port];branch=[branch];rport=5060 Contact: sip:[local_ip]:[local_port] To: sip:[field0]@[local_ip]:[local_port];tag=[call_number] From: 101sip:[field0]@[remote_ip];tag=[call_number] Call-ID: [call_id]@[remote_ip] CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO User-Agent: Sipp v1.1-TLS, version 20061124 Allow-Events: message-summary, dialog Content-Length: 0 ]] /send /scenario users.csv Description: Binary data -- _ -- 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 1.8.2.3 Now Available
On 01/26/2011 03:06 PM, Warren Selby wrote: Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new versioning methods made updates into 1.8.x releases and security updates into 1.8.x.y releases? Security fixes and regression fixes can cause sub-point releases. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 2:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 01:19 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 1:50 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt http://webmail.zktech.com/public/downloadfile.aspx?f=KERoF6PWf6e2FK8S5zgEDs 02rFGdd7zE0AIG7tjbCR9a06oFY1NwFap58zDWva3BcdOp%2b%2f%2fuBo8%3d cap-t38.pcap http://webmail.zktech.com/public/downloadfile.aspx?f=ulHIhepag5qoKm0cTUmljm T%2f7YCcOPvzlyZcnZg%2fG2B25W%2fsSr6Uwbu%2bET3kbKw84pTJjtuqrPQ%3d Unfortunately that log capture is incomplete; it doesn't include any of the messages that res_fax emits as it goes through T.38 negotiations. Please ensure that your 'console' channel in logger.conf has 'debug,verbose,warning,notice,error,fax' enabled and that you have 'core set verbose 10' and 'core set debug 10' set before the call attempt begins (or at least before ReceiveFAX is executed). If the server is only processing this particular call, then 'sip set debug on' would also be helpful. - Kevin I will get the additional debugs done when there is no other load on the fax. Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 Thanks Bryant -- _ -- 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 1.8.2.3 Now Available
On 11-01-26 04:07 PM, Kevin P. Fleming wrote: On 01/26/2011 03:06 PM, Warren Selby wrote: Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new versioning methods made updates into 1.8.x releases and security updates into 1.8.x.y releases? Security fixes and regression fixes can cause sub-point releases. A version bump from 1.8.2 to 1.8.3 would mean all changes since 1.8.2-rc1 was created would be included. A bump from 1.8.2 - 1.8.2.1 - 1.8.2.2 - etc... includes minor changes based on the base 1.8.2 version with very select fixes. Leif. -- _ -- 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] res_fax
On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 4:52 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Thanks Bryant -- _ -- 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
[asterisk-users] Asterisk 1.8.2.3 Now Available
Asterisk 1.8.2.3 Now Available From: Asterisk Development Team asteriskt...@digium.com To: Asterisk Development Team asteriskt...@digium.com Date: Today 18:18:28 The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by mnicholson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3 Thank you for your continued support of Asterisk! Un abrazo, helius -- Prefiro um engano que me divirta a uma experiência que me entristeca. -- William Shakespeare -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 5:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 04:16 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 4:52 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Most definitely; I can see cases like yours where someone would want to be able to forcibly disable T.38 for specific calls for troubleshooting purposes. In fact... if you give me about 15 minutes, I'll commit a patch to Asterisk trunk to add an option to do that, and you can backport it to the version you are using :-) Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? Much thanks on this. Bryant -- _ -- 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] res_fax
On 01/26/2011 04:36 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 5:21 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 04:16 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 4:52 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Most definitely; I can see cases like yours where someone would want to be able to forcibly disable T.38 for specific calls for troubleshooting purposes. In fact... if you give me about 15 minutes, I'll commit a patch to Asterisk trunk to add an option to do that, and you can backport it to the version you are using :-) Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] SIP, IAX2 and ISDN ISUP data
On 01/25/2011 12:44 AM, Phil Lello wrote: Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the voicemail DDI) - Originally Called Number (e.g. So if Adam phones Bob, Bob is diverted to Charlie, and Charlie is diverted to Voicemail, then Adam probably doesn't want Charlie's Voicemail). Asterisk 1.8 can receive, transmit and transport all this information over ISDN and SIP, including mid-call updates. I believe this information should be in SIP Divert headers, can someone confirm this? There are a number of SIP headers involved. Diversion, P-Asserted-Identity and Remote-Party-Id, if not others. Do I get the same information if I use an IAX2 connection to connect a local Asterisk server to an external one? It is possible that this information will transport properly across IAX2 connections between Asterisk 1.8 servers, but that scenario wasn't tested by the developers that worked on it. Does IAX2 route GSM/ISDN SMS between servers, and if so, would the remote/ISDN connected server need to explicitly support this, or do the remote cards look local? Asterisk does not support native SMS, and doesn't transport it between servers. There is an SMS application, but it is an SMS endpoint, not a router. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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
[asterisk-users] SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?
Hi Everyone, I want to call first party using a .callfile and a second party using a context and then bridge the two calls. I MUST make sure that first party picks up first and then the second party should be dialed. Trying the following using an internal extension works nicely and the playback file is play after the extension picks up. But using the same method for calling an outside phone number (using a good quality SIP provider) does not wait for the channel to come up and starts the Playback line right away. What is the fault behind this and what is workaround? This works: *originate sip/101 extension s@dial_wait* [dial_wait] exten = s,1,Answer exten = s,n,Playback(Please_wait_as_dial_the_second_party) exten = s,n,NoOp(Calling second party) exten = s,n,Dial(SIP/sip_provider/1214555) This doesn't wait for channel to come up and jumps to Playback (s,2) without even the first party yet picking up: *originate SIP/sip_provider/1214888 extension s@dial_wait* * * *Thanks,* -- _ -- 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
[asterisk-users] Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes answeronpolarityswitch=yes usecallerid=yes cidsignalling=bell cidstart=ring ;hidecallerid=yes ;hidecalleridname=yes ;waitfordialtone=yes ;mwimonitor=no ;mwilevel=512 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ;mwisendtype=rpas,lrev callwaiting=yes ;restrictcid=no usecallingpres=yes sendcalleridafter = 1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=no group=1 callgroup=1 pickupgroup=1 ;immediate=yes immediate=no callerid = asreceived useincomingcalleridondahditransfer = yes callprogress=yes progzone=us faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no faxbuffers=6,full -- _ -- 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
[asterisk-users] Asterisk 1.8 and Cisco 7920
I was curious to know if anyone has had any luck getting Cisco phones working with Asterisk and chan_skinny? Specifically, a Cisco 7920. If a SIP firmware was available, I'd just use that. It works fine with chan_sccp and 1.6, but my understanding is that chan_sccp does not work with 1.8 yet. My workaround for the moment is a 1.6 VM tied via IAX to my 1.8 box. This will work for now, but it's a lot uglier than I prefer. Thanks for your help. -- _ -- 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] res_fax
Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 - Kevin I downloaded 1.8.2.3 and copied the modified version of res_fax.c into my the res folder. I built and installed the version of asterisk. When I use the new n option with ReceiveFAX I get a bunch of WARNING messages on the console that state. [Jan 26 20:43:38] WARNING[23393]: chan_sip.c:6047 sip_write: Asked to transmit frame type slin, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) If I shut of the n option it goes back to the normal behavior. It appears that there is somthing missing in the n option and it is not causing it to fall back to audio only mode. as it would if t38pt_udptl=no Bryant -- _ -- 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 1.8.2.3 Now Available
On 1/26/2011 3:18 PM, Asterisk Development Team wrote: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by mnicholson) I can confirm that this resolves the issue I was having. Thanks to all who were involved, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Recommended Windows client to display CID?
Gilles skrev: Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. Hi Gilles, If you want someting really light weight there is always the old winpopup protocoll. For example: smbclient -M NETBIOSNAME text.txt (on linux) I am unsure about support in later versions of Windows but up to at least win2000 this pops up a simple message box on the windows machine remotely. /Magnus -- _ -- 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
