Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
My asterisk server(s) are behind NAT, and I am a customer of Vonage (thrice-over), iconnecthere, and Net2Phone. There are still some rough edges (especially with iconnecthere) but overall it is not correct to say that they won't work. B. Thats great to hear. Can you please share your config files that connect iconnecthere and net2phone via SIP? I think there are a number of people here who have tried and not been able to get it to work. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
rnc Info Lists wrote: Thats great to hear. Can you please share your config files that connect iconnecthere and net2phone via SIP? I think there are a number of people here who have tried and not been able to get it to work. Here's what I'm using for iconnecthere. They provide me with both origination and termination, btw, so there are clauses that handle each. *** in sip.conf: register = 18005551212:[EMAIL PROTECTED] (first part is my inbound phone number, second is account password) [iconnect] type=peer username=12312312 secret= callerid = My Name 18005551212 host=213.137.73.140 And in extensions.conf: exten = _11.,1,Goto,iconn|BYEXTENSION|1 Later on. . . [iconn] exten = _11NXXNXX,1,StripMSD,1 exten = _1NXXNXX,2,Prefix, exten = _1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]||r For origination: exten = 15126919417,1,Dial,SIP/ata1|23 Note I'm using the old (deprecated) syntax for the various commands. And I don't pretend this is beautiful or optimal syntax. The preceding the number was something they told me to use to get gsm encoding. FWIW. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Dev Kit Lite
costas wrote: I installed Asterisk as per instructions in the FAQ on the digium.com site. Double checked it. I also think they have a bug in the zapata.conf where the context should be incoming and not internal. 1) I hear no dialtone when I pickup the phone on the S100U. Asterisk sees the event and displays the message on the screen. I tried dialing but nothing happens. I hangup and * shows the hangup event. I did set the phone.conf mode to be dialtone but nothing works. 2) I can dial from the outside and the phone on the S100U rings. Asterisk shows Ringing. However when I pick it up it says Attempting to Bridge. I wait a while but doesnt work. There is no communication between the outside phone and the one on the S100U. There is no sound card in the machine. I assume its not the problem. The machine is an RH9 with 2.2Ghz and 256RAM. Any ideas or help would be appreciated. I had similar problems with my S100U.. In fact I went through three of them in 6 months... The just kept breaking.. So I would be inclined to suspect a faulty S100U over an incorrect configuration.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold with Voicetronix
I have now is music on hold I have installed ztdummy MOH gives me this message now Read 372 bytes of audio while expecting 1600 And no sound If I run modprobe ztdummy I get no sound Even the welcome message etc do not work And the error message changes to Read 372 bytes of audio while expecting 625 Any ideas Regards Mick West ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail help
hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to call.. something like that. i need some help here. also, is there any way to find demo sound files? thank you. cm = Designs __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I am using [EMAIL PROTECTED], all the SIP traffic will be sent to
iptel.org proxy and the proxy will take care of NAT traversal. Currently
I forward all numbers begining with 3 to iptel.org beucase I don't know
how to create fall-back rule that will match when there are no other
rules (neither i nor _. works for me).
In the other direction, calls to [EMAIL PROTECTED] get translated to
[EMAIL PROTECTED] and user jan registered at the asterisk box will
receive them.
To able able to call anywhere through iptel.org, From header field must
contain iptel.org so fromdomain parameter is necesarry in [iptel]
section.
Testing scenario was as follows:
[Caller][*]---[NAT][iptel.org (public inet)][NAT]---[Callee]
and vice versa.
sip.conf and extensions.conf follow. I have no previous experience in
configuriing asterisk so maybe the config files are not the best ones, I
simply took John Todd's config files and tweaked them a bit, it seems to
work for me.
To iptel.org proxy asterisk looks like a normal SIP user agent behind
NAT. iptel.org is running SER with extended nathelper and RTP proxy.
Jan.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = from-sip ; Default for incoming calls
;
register = asterisk:[EMAIL PROTECTED]/jan ; Register with a SIP
provider
[iptel]
type=friend
username=asterisk
secret=password
fromdomain=iptel.org
host=iptel.org
[jan]
type=friend
username=jan
host=dynamic
canreinvite=no
extensions.conf:
[from-sip]
exten = jan,1,Dial(SIP/jan)
exten = jan,2,Hangup
exten = _3.,1,SetCallerID(jan)
exten = _3.,2,SetCIDName(Jan Janak)
exten = _3.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _3.,4,Playback(invalid)
exten = _3.,5,Hangup
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RE: [Asterisk-Users] Nextone softswitch testing and Asterisk long distance
Hi, I would love to participate in your test. We have several * machines. Please let me know further details. Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_mysql.so
Can anyone give me presise instructions on how to compile cdr_mysql.so? When I initially installed asterisk on the system, I didn't have mysql installed. Since then I have installed mysql, created the database and table structure for cdr_mysql and placed the appropriate settings in the cdr_mysql.conf file. However when I do a show modules at the CLI I cannot find cdr_mysql.so. I've also noticed there's no line for it in the module section along with the fact that I cannot find the file cdr_mysql.so anywhere on the system. I've read the cdr_mysql.txt file in the asterisk doc directory but it doesn't say anything regarding how to compile. Thanks in advance for any suggestions/assistance, etc. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
[EMAIL PROTECTED] wrote: Can anyone give me presise instructions on how to compile cdr_mysql.so? When I initially installed asterisk on the system, I didn't have mysql installed. Since then I have installed mysql, created the database and table structure for cdr_mysql and placed the appropriate settings in the cdr_mysql.conf file. However when I do a show modules at the CLI I cannot find cdr_mysql.so. I've also noticed there's no line for it in the module section along with the fact that I cannot find the file cdr_mysql.so anywhere on the system. I've read the cdr_mysql.txt file in the asterisk doc directory but it doesn't say anything regarding how to compile. Thanks in advance for any suggestions/assistance, etc. AJ MySQL support for CDR has been moved out of the asterisk module in the CVS for licence reasons.. Using the same CVS server checkout the asterisk-addons module which is where the MySQL CDR stuff is now located.. and then run make install from there.. And if you are going to aske is the Voicemail2 MySQL supports is going to be removed, I am afraid I don't have an answer for you.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
On Sat, 25 Oct 2003 [EMAIL PROTECTED] wrote: Can anyone give me presise instructions on how to compile cdr_mysql.so? did you get asterisk-addons ? cdr_mysql been moved there When I initially installed asterisk on the system, I didn't have mysql installed. Since then I have installed mysql, created the database and table structure for cdr_mysql and placed the appropriate settings in the cdr_mysql.conf file. However when I do a show modules at the CLI I cannot find cdr_mysql.so. I've also noticed there's no line for it in the module section along with the fact that I cannot find the file cdr_mysql.so anywhere on the system. I've read the cdr_mysql.txt file in the asterisk doc directory but it doesn't say anything regarding how to compile. Thanks in advance for any suggestions/assistance, etc. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
I just went over and grabed the asterisk-addons directory from the CVS, changed into the directory and executed a make install and got the following error: make: ***[cdr_addon_mysql.o] Error 1 You have any suggestions about this? AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
[EMAIL PROTECTED] wrote: I just went over and grabed the asterisk-addons directory from the CVS, changed into the directory and executed a make install and got the following error: make: ***[cdr_addon_mysql.o] Error 1 You have any suggestions about this? AJ I just did a cvs update and recompiled it and it worked fine.. Have you got the mysql and mysql-devel packages installed? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 stops asterisk in the background
Please try a modification in /etc/rc.d/init.d/asterisk: Replace: daemon /usr/sbin/asterisk with daemon screen -d -m asterisk -vvvcng Hope this helps. Gus - Original Message - From: Alejandro Ruiz [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 24, 2003 6:11 PM Subject: [Asterisk-Users] G729 stops asterisk in the background I hope somebody can help me out... since I installed the G729 from digium, my asterisk box doesn't run in the background any more. I can only make it work with asterisk -vvvc; If I ran asterisk and then asterisk -rvvvc to see what's happening; once evrey module finished loading, the program stops; just as if I entered stop now does someone have any clue about it? thanks, alejandro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Manually Answer
[inbound-home]
exten = s,1,Dial(${PHONE3}${PHONE4},15)
Thanks Rich, this worked like a charm, I don't know why I was thinking in
reverse -- that I would have to have Asterisk answer it to pass the ringing
to the SIP phone or I would have to force a pickup with some sort of key
sequence.
Thanks!
Ben Wern
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RE: [Asterisk-Users] Compiling gastman under Win32
Where is the current Java binary? For that matter, where is the source for the Java version? Cheers, Steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Victor Medrano Sent: Friday, October 24, 2003 10:34 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling gastman under Win32 Download binary with java , works fine with 2000 + Xp regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol Sent: Friday, October 24, 2003 4:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Compiling gastman under Win32 Can anybody tell me what I need to have in order to make/compile gastman on Windows using VC++? Or do I want to download and install gcc for Windows? I have never worked with anything designed to be cross-platform compilable. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone using sipcall.co.uk ? Now sipphone
On Fri, 2003-10-24 at 21:12, David J Carter wrote: Thanks Dave, I can now call a sipphone number from * but get no voice throughput. I still don't see anything coming in from sipphone though. Dave Have a look at rtp.conf I have 8000 - 8060 there. by default its 1, if you have a Grandstream you must tell it what to use. It works because sipphone newbies keep calling me. They can't believe transatlantic is so good. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: I just went over and grabed the asterisk-addons directory from the CVS, changed into the directory and executed a make install and got the following error: make: ***[cdr_addon_mysql.o] Error 1 You have any suggestions about this? AJ I just did a cvs update and recompiled it and it worked fine.. Have you got the mysql and mysql-devel packages installed? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Install the zlib and zlib devel packages On Sat, 2003-10-25 at 12:18, [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: I just went over and grabed the asterisk-addons directory from the CVS, changed into the directory and executed a make install and got the following error: make: ***[cdr_addon_mysql.o] Error 1 You have any suggestions about this? AJ I just did a cvs update and recompiled it and it worked fine.. Have you got the mysql and mysql-devel packages installed? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Yes I do have gzip installed on that box. Any other ideas? On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Thanks a bunch On Sat, 25 Oct 2003, Eric Wieling wrote: Install the zlib and zlib devel packages On Sat, 2003-10-25 at 12:18, [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: I just went over and grabed the asterisk-addons directory from the CVS, changed into the directory and executed a make install and got the following error: make: ***[cdr_addon_mysql.o] Error 1 You have any suggestions about this? AJ I just did a cvs update and recompiled it and it worked fine.. Have you got the mysql and mysql-devel packages installed? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Dev Kit Lite
costas wrote: How did you determine that they are faulty? Did Digium replace them? Thanks Well when I started the device worked and over time the phone started crackling and then randomly stopped providing dial tone and then stopped providing dial tone at all.. I am in the UK so I contacted the agent that I got it from and it was swapped out.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
gzip is not zlib. On my Mandrake 9.2 system the zlib packages are: zlib1-1.1.4-8mdk zlib1-devel-1.1.4-8mdk On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
On Sat, 25 Oct 2003 [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? You are looking for the the libz stuff, which if you use RedHat is a part of zlib-devel-1.1.3-25.7 (or whatever the right number is for your distribution). You should be able to see it in somewhere like /usr/lib/libz.a or /usr/lib/libz.so.1.1.3 Michael On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Thanks a lot your earlier suggestion worked. The system was lacking zlib-devel. Now where do I insert the lines for it to load the cdr_mysql.so since I have it built? Can you give me an exact example of what to put here? AJ On Sat, 25 Oct 2003, Eric Wieling wrote: gzip is not zlib. On my Mandrake 9.2 system the zlib packages are: zlib1-1.1.4-8mdk zlib1-devel-1.1.4-8mdk On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS update
make update, not make upgrade On Sat, 2003-10-25 at 13:33, Rich Adamson wrote: In attempting to make my asterisk server the latest and the greatest tonight I attempted to upgrade my CVS. In the asterisk directory I ran the make clean, executed the CVS update -d, and after all the files completed ran make upgrade. My problem is that when I pull up the CLI the cvs version that is showing is the same date as my initial install. Does this mean that the upgrade did not go correctly? Or does the number just never change after the initial install? Everything seems to be working ok after I restarted the server. I would just like to be sure I'm using the newest CVS that I tried to get. In my /usr/src/asterisk dir, all I do is: make upgrade -- it connects to CVS and shows me all the new files it downloaded make clean ; make install I just tried the above, and absolutely nothing was updated. I actually checked *.c files for dates/times, and all were from previous cvs. Anything special one needs to do leading up to issuing the above? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS update
On Sat, 25 Oct 2003, Rich Adamson wrote: make upgrade -- it connects to CVS and shows me all the new files it downloaded make clean ; make install I just tried the above, and absolutely nothing was updated. I actually checked *.c files for dates/times, and all were from previous cvs. Anything special one needs to do leading up to issuing the above? It is possible you have a sticky tag or date set on your CVS checkout. Try taking a look at the output of: cvs status Makefile Or any other file. To get rid of the sticky tag you would want to do: cvs update -d -A Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Modify /etc/asterisk/modules.conf load = cdr_mysql.so or load = cdr_addon_mysql.so Then, modify your cdr_mysql.conf, like this: [global] hostname=localhost dbname=astcdr password=12jaslap3 user=amadata sock=/var/lib/mysql/mysql.sock Hope this helps. Gus - Original Message - From: [EMAIL PROTECTED] To: Eric Wieling [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Saturday, October 25, 2003 2:58 PM Subject: Re: [Asterisk-Users] cdr_mysql.so Thanks a lot your earlier suggestion worked. The system was lacking zlib-devel. Now where do I insert the lines for it to load the cdr_mysql.so since I have it built? Can you give me an exact example of what to put here? AJ On Sat, 25 Oct 2003, Eric Wieling wrote: gzip is not zlib. On my Mandrake 9.2 system the zlib packages are: zlib1-1.1.4-8mdk zlib1-devel-1.1.4-8mdk On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr_mysql.so
- Create a conf file called cdr_mysql.conf in /asterisk/ [global] hostname=localhost dbname=asterisk password=somepass user=someuser - Add load = cdr_addon_mysql.so to /asterisk/modules.conf If you are getting errors with mysql cdr while loading asterisk, check that you can connect to MySQL from a shell (mysql -u username -p) or from MySQL Front by entering the login info for your asterisk db user. You may want a MySQL client so you can view your cdr details until you get a frontend going (astweb ??). John Haigh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Saturday, October 25, 2003 1:58 PM To: Eric Wieling Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] cdr_mysql.so Thanks a lot your earlier suggestion worked. The system was lacking zlib-devel. Now where do I insert the lines for it to load the cdr_mysql.so since I have it built? Can you give me an exact example of what to put here? AJ On Sat, 25 Oct 2003, Eric Wieling wrote: gzip is not zlib. On my Mandrake 9.2 system the zlib packages are: zlib1-1.1.4-8mdk zlib1-devel-1.1.4-8mdk On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS update
Opps, funny how the fingers do the walking following the eyes, without questioning the print... ;) make update, not make upgrade On Sat, 2003-10-25 at 13:33, Rich Adamson wrote: In attempting to make my asterisk server the latest and the greatest tonight I attempted to upgrade my CVS. In the asterisk directory I ran the make clean, executed the CVS update -d, and after all the files completed ran make upgrade. My problem is that when I pull up the CLI the cvs version that is showing is the same date as my initial install. Does this mean that the upgrade did not go correctly? Or does the number just never change after the initial install? Everything seems to be working ok after I restarted the server. I would just like to be sure I'm using the newest CVS that I tried to get. In my /usr/src/asterisk dir, all I do is: make upgrade -- it connects to CVS and shows me all the new files it downloaded make clean ; make install I just tried the above, and absolutely nothing was updated. I actually checked *.c files for dates/times, and all were from previous cvs. Anything special one needs to do leading up to issuing the above? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql.so
Thanks, got it working, Thanks to everyone! AJ On Sat, 25 Oct 2003, CW_ASN wrote: Modify /etc/asterisk/modules.conf load = cdr_mysql.so or load = cdr_addon_mysql.so Then, modify your cdr_mysql.conf, like this: [global] hostname=localhost dbname=astcdr password=12jaslap3 user=amadata sock=/var/lib/mysql/mysql.sock Hope this helps. Gus - Original Message - From: [EMAIL PROTECTED] To: Eric Wieling [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Saturday, October 25, 2003 2:58 PM Subject: Re: [Asterisk-Users] cdr_mysql.so Thanks a lot your earlier suggestion worked. The system was lacking zlib-devel. Now where do I insert the lines for it to load the cdr_mysql.so since I have it built? Can you give me an exact example of what to put here? AJ On Sat, 25 Oct 2003, Eric Wieling wrote: gzip is not zlib. On my Mandrake 9.2 system the zlib packages are: zlib1-1.1.4-8mdk zlib1-devel-1.1.4-8mdk On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote: Yes I do have gzip installed on that box. Any other ideas? On Sat, 25 Oct 2003, WipeOut wrote: [EMAIL PROTECTED] wrote: Yes I do have the mysql and mysql-devel packages installed. With my very limited knowledge of C/C++ here's what seems to be the culpret line right before the error: /usr/bin/ld: cannot find -lz Any suggestions here? AJ I think thats gzip.. Have you got gzip installed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup (*8) on SIP devices.
Just submitted a patch for this on asterisk-dev. Quick fix add the following line above line 5022 in chan_sip.c ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere as was indicated possible earlier. Am getting the following on the Asterisk console: -- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] == No one is available to answer at this time sip.conf is: [delta3] type=peer username= secret= host=213.137.73.140 the extension.conf entry is: exten =_1706NXX,1,Dial,SIP/[EMAIL PROTECTED] Am I missing something?? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail.conf in MySQL is not functioning
Voicemail.conf in MySQL is not functioning where I get the following error
from Asterisk messages log file:
CLI debug output is as follows:
Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack
-- Playing 'vm-login'
-- Playing 'vm-password'
-- Incorrect password '1234' for user '0' (context = any)
-- Playing 'vm-incorrect'
-- Playing 'vm-password'
-- Incorrect password '2421' for user '2205' (context = any)
-- Playing 'vm-incorrect'
Here are my configs
In extensions.conf I am using Voicemail2 and VoiceMailMain2 that has support
for MySQL
exten = 8500,1,VoiceMailMain2
In voicemail.conf I have the MySQL connectivity settings in [general]
dbhost=localhost
dbname=asterisk
dbuser=someuser
dbpass=somepass
I have commented out the entire [default] section and it's mailboxes.
I do have MySQL working with CDR MySQL from asterisk-addons
thanks,
John Haigh
[Asterisk-Users] X100P stopped working
I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I rebooted a few times too, to make sure everything had been cleared out. === [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074404064]: File chan_zap.c, Line 6986 (load_module): Ignoring rxwink WARNING[1074404064]: File chan_zap.c, Line 626 (zt_open): Unable to specify channel 1: No such device or address ERROR[1074404064]: File chan_zap.c, Line 4949 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074404064]: File chan_zap.c, Line 6730 (load_module): Unable to register channel '1' WARNING[1074404064]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074404064]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I rebooted a few times too, to make sure everything had been cleared out. === [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074404064]: File chan_zap.c, Line 6986 (load_module): Ignoring rxwink WARNING[1074404064]: File chan_zap.c, Line 626 (zt_open): Unable to specify channel 1: No such device or address ERROR[1074404064]: File chan_zap.c, Line 4949 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074404064]: File chan_zap.c, Line 6730 (load_module): Unable to register channel '1' WARNING[1074404064]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074404064]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I'm probably not going to be of much help since I'm rather new at this as well. I do have two x100p's running on RH9 though. Check your /etc/zaptel.conf to ensure you have something like: fxsks=1 loadzone=us Might try: /usr/src/zaptel/ztcfg -vv If I run /sbin/lsmod, I see: Module Size Used byNot tainted snip wcfxo 9056 2 zaptel180128 8 [wcfxo] ppp_generic2 0 [zaptel] snip Not sure about this, but using rmmod, insmod and lsmod to unload and reload might help as the running (old) modules vs newly compiled modules might be causing some sort conflict. (Out of my league.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Did you try to use *8 only instead of *8# ? Last time when I tried *8 picked the call with known results but I haven't tested any patches yet. I really hope call pickup now works. -- Pertti Rich Adamson wrote: Just submitted a patch for this on asterisk-dev. Quick fix add the following line above line 5022 in chan_sip.c ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk External Resources Page
I've submitted http://bugs.digium.com/bug_view_page.php?bug_id=434 requesting that Digium put up a page with links with external Asterisk related resources. If you have a web site with Asterisk related information, patches, samples, documentation, etc, please add a bugnote to the above URL. There is a lot of good information out there, but time and time again I hear complaints that nobody can find it. --Eric -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:SIP Provider Question
David, We has successfully tested last week interoperability between Asterisk and our SIP softswitch. We can definately help you with your project. Our company is New York based, so it will be very easy to get interconnection with you. We are mostly in wholesale business. Also you can get ,if needed, direct interconnection at our second POP in Zurich, Switzerland. Please contact me , if any interest. Alexander Kandelaki Stealth Telecommunications New York, NY Michael Bielicki [EMAIL PROTECTED] Wed, 1 Oct 2003 20:25:59 +0200 Previous message: [Asterisk-Users] SIP Provider Question Next message: [Asterisk-Users] grandstream phones and Transfer Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] We can do that in the UK and in Poland and soon in more countries, but since you are US based I would recommend rather to talk to Jeremy at Nufone. On Wednesday 01 October 2003 8:30 pm, David Harris wrote: Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly asterisk box to the SIP provider. Also I would need to able to port the companies existing numbers over to the SIP provider in order receive incoming calls. Thanks, David -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender.
Re: [Asterisk-Users] X100P stopped working
I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I rebooted a few times too, to make sure everything had been cleared out. === [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074404064]: File chan_zap.c, Line 6986 (load_module): Ignoring rxwink WARNING[1074404064]: File chan_zap.c, Line 626 (zt_open): Unable to specify channel 1: No such device or address ERROR[1074404064]: File chan_zap.c, Line 4949 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074404064]: File chan_zap.c, Line 6730 (load_module): Unable to register channel '1' WARNING[1074404064]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074404064]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Just a thought... You did do a make clean first before recompiling? Couldn't tell from message if you just updated the source for asterisk or everything? The reason I ask because when I do updates I update everything ie. 1st, make clean,update Zaptel, make, make install 2nd, make clean, update Libpri, make, make install 3rd, make clean, update Asterisk, make, make install At least that's how I would do it, I believe asterisk relies on some shared libs when compiling and I want to make sure everything is matched up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Confuson on iax calls (register or not?)
Think I'm a little confused on registering an iax connection; could
someone enlighten me?
I guess the real question is... when two * machines are going to rely
on an iax link (each with their own dial plan), do both machines have
to register with each other (eg, both need a 'register' statement)?
Or, will a single machine doing the registering cause the opposite
machine to recognize the registration, and allow calls to be originated
in both directions?
If so, assume machine B registers with machine A (machine B has the
register statement). Then, in machine A's extensions.conf dial plan,
what would the statement similar to
exten = _6X.,1,Dial(IAX/npi-off:[EMAIL PROTECTED]/${EXTEN-1})
look like? (I'm assuming the above might use the context for which
machine B registered with?)
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Re: [Asterisk-Users] Looking for a 12 to 24 FXO Channel Bank in Colombia
I have spotted some candidates at EBay... Problem is that Pay Pal cannot handle payments from my country, Colombia. Paypal isn't the only way to pay -- a U.S. Money order or wire transfer should work just fine... Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More beginner questions...
Questions ... OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes, and extensions. All exciting. Two questions: I'm in a natted environment and need to utilize a SIP provider to make calls in the US. Currently I have Vonage in my natted network and it works fine, however I understand there is no real way to make asterisk talk to Vonage because they have a closed system. So, the question is, what SIP provider do I go with? Second, I have 1 phone line coming into the house and I would like it to be routed through Asterisk. Is my best choice for this, a modem? Or, is there other hardware I should consider? Regards, Phillip -- Phillip Jackson - [EMAIL PROTECTED] President, The Jackson Group - Intelligent IT. (TM) Ph 410.320.2138 Fx 443.321.8713 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More beginner questions...
Questions ... OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes, and extensions. All exciting. Two questions: I'm in a natted environment and need to utilize a SIP provider to make calls in the US. Currently I have Vonage in my natted network and it works fine, however I understand there is no real way to make asterisk talk to Vonage because they have a closed system. So, the question is, what SIP provider do I go with? Does it have to be SIP? Several providers will do this over IAX...nufone.net, voicepulse.com Second, I have 1 phone line coming into the house and I would like it to be routed through Asterisk. Is my best choice for this, a modem? Or, is there other hardware I should consider? X100P. Consider nothing else. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More beginner questions...
Is IAX difficult to configure? Do you have sample configs I could look at? Is there a rec IAX provider? Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
