RE: [Asterisk-Users] zaptel wont compile
I think we covered this not too long ago. Did you make sure your link /usr/src/linux-2.4 is linked to /lib/modules/2.4XXX/build (replacing the X's with your corresponding kernel)? If you need to know your kernel, it is usually posted on the login screen or type `uname -r` for the kernel version. HTH Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] -Original Message- From: AJ Grinnell [mailto:[EMAIL PROTECTED] Sent: Thursday, August 12, 2004 8:50 AM To: Asterisk Subject: [Asterisk-Users] zaptel wont compile Trying to update to the latest cvs, but Asterisk complained that zaptel was too old. Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB - I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-point er -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c zaptel.c cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB - I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-point er -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:61: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:61: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:67: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:67: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:67: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:68: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:68: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:71: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:71: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:72: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:72: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:73: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:73: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:73: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:73: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:75: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:75: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:76: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:76: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:76: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:76: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:78: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:78: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:79: `sscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:79: warning
RE: [Asterisk-Users] Asterisk MyPhoneCompany.com (aka Talk(n))
I signed up for BV and entered everything, to the best of my knowledge, in the sip.conf and extensions.conf file, but am having problems getting it to connect. If someone could show me an example of their working BV sip.conf that would be greatly appreciated. I am certain that I have the extensions.conf correct since I can see it in the console (see below). -- parse_srv: SRV mapped to host proxy.lax.broadvoice.com, port 5060 -- Executing SetCallerID(SIP/loni-5ddf, 4047952206) in new stack -- Executing Dial(SIP/loni-5ddf, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] Aug 12 09:06:16 WARNING[-163574864]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing Congestion(SIP/loni-5ddf, ) in new stack == Spawn extension (from-sip, 13342156551, 3) exited non-zero on 'SIP/loni-5ddf' I see these WARNING messages when I go to make the call (that is to a line that matches those dialing plans). I have made certain that I am using the correct username/passwords that were sent to me in the activation email. Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 7:04 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk MyPhoneCompany.com (aka Talk(n)) BroadVoice DOES work, if you sign up with their BYOD plan, and you select other device it dosen't require you to enter a MAC... Several people in the list including myself use BroadVoice with *. I am very happy with it and I'd be more than willing to help you set it up if you need it! :) Also many have reported that IconnectHere works with *, a while ago people were complaining of DTMF problems I don't know if this was ever resolved. Obviously without inbound DTMF, using * would be pretty useless... I just wanted to know if people have tried some of these Other Guys as * providers, I'm trying to come up with a list of providers (SIP and IAX) that work well with *... I kinda want to make a handbook of sorts for some friends who want to play with *... -Chris - Original Message - From: Travis Conway [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 4:51 PM Subject: RE: [Asterisk-Users] Asterisk MyPhoneCompany.com (aka Talk(n)) I looked at BroadVoice and thought about doing it, but does * actually have a MAC address? And do they mean the MAC address of my * box's NIC? We have a router between the * and the internet, do they want the MAC of my router? Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 6:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk MyPhoneCompany.com (aka Talk(n)) They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] StanaPhone and Asterisks
I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com fromdomain=sip.stanaphone.com insecure=very nat=yes extensions.conf: exten = _1NX,1,SetCallerId,3475341914 exten = _1NX,2,Dial(SIP/[EMAIL PROTECTED]) exten = _1NX,3,Playback(invalid) exten = _1NX,4,Hangup When it calls it goes to playing invalid and on my console I see where they gave me an error 503 Service Unavailable but when I connect directly to StanaPhone using a SIP softphone, I can dial out with no problems. Do any of you use stanaphone and happen to have some configs that work? Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] StanaPhone and Asterisks
Yeah, if you able to get it to work. I have been working on this for about 3 hours now. I can't get the thing to connect. I was once getting 503 now I am getting 403 Forbidden from StanaPhone. Ergh, this is very annoying. Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 6:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] StanaPhone and Asterisks - Original Message - From: Pulu 'Anau [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Travis Conway [EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 3:52 PM Subject: Re: [Asterisk-Users] StanaPhone and Asterisks From my sip.conf: register=9146186144:[EMAIL PROTECTED]/9146186144 [stanaphone] fromdomain=sip.stanaphone.com fromuser=9146186144 type=peer secret=notyoursecret auth=MD5 username=9146186144 host=sip.stanaphone.com dtmfmode=info context=default canreinvite=no disallow=all allow=ulaw allow=ilbc It looks like the only big difference is with the fromuser thing. I had a really hard time with it, and ended up searching alot until I found this setup in some cablemodem fansite. Outgoing always seems to work fine, incoming I have problems with, sometimes it fails on the registration with unauthorized errors. Pulu Neat, another *-capable provider... The insecure=very should fix the incoming calls issue if it's on *'s end, I noticed that you didn't have that anywhere. Also adding an [incoming] context as type peer without username and password info should work as well... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MyPhoneCompany.com (aka Talk(n))
I looked at BroadVoice and thought about doing it, but does * actually have a MAC address? And do they mean the MAC address of my * box's NIC? We have a router between the * and the internet, do they want the MAC of my router? Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 6:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk MyPhoneCompany.com (aka Talk(n)) They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Does IConnectHere still work with asterisk?
Couldn't you get around this by changing the SIP useragent variable? Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] -Original Message- From: todd palumbo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 11, 2004 5:51 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Does IConnectHere still work with asterisk? I want to test outgoing VoIP Cheap, and they have a $5.95/month plan that would be perfect. I want it to interface through asterisk though, and I heard some providers are blocking asterisk. I just want to make sure that it still works before I sign up. Also, if you have any experience with them, let me know. Thanks, -Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Application asterisk uses obsolete OSS audio interface
Should I be concerned about this? It seems to only happen when my MoH switches songs. The songs sound as good as an 8k/s song would. Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED]
[Asterisk-Users] CVS download
I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S- server_register(fpm-world-mix.mp3, 1.1, , , , , ) S- Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme
I am trying to get a simple MeetMe running between
a few SIP phones here in our office.
Here is a clip from my extensions.conf
exten = 9000,1,Ringingexten =
9000,2,Answerexten = 9000,3,Wait(1)exten
= 9000,4,MeetMe(|Md)
Here is what my console says:
-- Executing
Ringing("SIP/loni-b550", "") in new stack -- Executing
Answer("SIP/loni-b550", "") in new stack -- Executing
Wait("SIP/loni-b550", "1") in new stackAug 6 13:48:56
WARNING[-298230864]: pbx.c:1257 pbx_extension_helper: No application 'MeetMe'
for extension (from-sip, 9000, 4) == Spawn extension (from-sip, 9000,
4) exited non-zero on 'SIP/loni-b550'My meetme.conf file looks correct
(nothing really in it, just a [rooms] block).
Perhaps meetme isnt installed? I used an RPM
for RC-1 I found on a mirror.
--Travis Conway[EMAIL PROTECTED]FWD:
414668+1 334 220-7519 (T-Mobile)
Re: [Asterisk-Users] Re: meetme
I just downloaded the new stuff form CVS and compiled it, but cannot find the meetme so file. What gives? -- Travis Conway [EMAIL PROTECTED] FWD: 414668 +1 334 220-7519 (T-Mobile) - Original Message - From: Tony Mountifield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 06, 2004 2:26 PM Subject: [Asterisk-Users] Re: meetme In article [EMAIL PROTECTED], Travis Conway [EMAIL PROTECTED] wrote: Aug 6 13:48:56 WARNING[-298230864]: pbx.c:1257 pbx_extension_helper: No application 'MeetMe' for extension (from-sip, 9000, 4) Check that you have the file /usr/lib/asterisk/modules/app_meetme.so and that /etc/asterisk/modules.conf has a [modules] section with either: * A line saying autoload=yes and NO line saying noload = app_meetme.so * A line saying load = app_meetme.so Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] users
Hello Guys, I just setup an Asterisk server here at work and have just a question that I was hoping you could help me with. How do I run asterisk as a user other than root? It seems that if I try to start it as a user I created it doesnt actually start.--Travis Conway[EMAIL PROTECTED]FWD: 414668+1 334 220-7519 (T-Mobile)
