[Asterisk-Users] RealTime Drivers Connectivity Error
Hello *'s, i am using Realtime Sip drivers but its not working here is my configs: extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends = mysql,asterisk,sip_friends res_mysql.conf [general] dbhost = localhost.localdoamin/127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = 123456 dbport = 3306 dbsock = /var/lib/mysql/mysql.sock error detail: Dec 31 01:20:49 ERROR[4298]: res_config_mysql.c:617 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on localhost.localdomain/127.0.0.1. Check debug for more info. == Registered application 'UserEvent' [app_verbose.so]Segmentation fault (core dumped) i change dbhost parameter several times like(localhost,192.168.10.193 etc) but can't works I am using latest CVS-Head kindly pointout my mistakes. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] how to debug frame slips?
Hi, I have had problems with receiving faxes with rxfax. Using bristuff with four hfc cards and spandsp 0.2.0 pre6 I got the symptoms of frame slips. I've tried all the debugging tips in this thread. I've tried moving from kernel 2.4 to 2.6.7. The only symptom of something wrong has been the 32k interrupts per second generated by the zaphfc driver. All the time the sound quality when listening has been good. Applying the patch from Florian Zumbiehl to zaphfc seems to have solved my problems. The number of generated interrupts when the system was idle decreased drastically. Thanks Florian! Link to the patch can be found at the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc /Nils On Mon, 27 Dec 2004, Joe Presto wrote: Hi, I'm running into issues receiving faxes which, from what I have read, may be caused by frame slips. While I can find many posts saying to investigate it, I can't find any that describe *how* to debug the problem. Tried searching this list as well to no avail. Any pointers would be greatly appreciated. FYI, I'm running wbel, AMP 1.04, spandsp 2pre4. Faxing to a pstn on a t400p. Most faxes don't get received at all, but those that do have what appear to be the traditional signs: horizontal lines, most of the page cut off, etc.. And here's a basic linux question as a followup: if I try different hardware, can I use the existing build - or do I need to rebuild linux to properly detect the changed devices? Thanks in advance - Joe Presto Nils Segerdahl --- Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03 http://www.upsys.seFax: (+46) (0)18 56 80 49 --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardened gentoo (selinux) asterisk problem
Hey, I've just installed hardened gentoo with selinux and emerged the selinux policy's for asterisk and emerged asterisk after it, now whenever i want to run asterisk i get: Dec 31 11:56:46 WARNING[4248]: manager.c:1474 init_manager: Unable to bind socket: Cannot assign requested address and in my sip.conf i have: port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=10.2.1.2 ; IP address to bind to (0.0.0.0 binds to all) wich should be correct, i also tried 0.0.0.0 but same error and when using netstat -a it doesn't show anything else that uses up that port. anyone an idea? grtz -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] attachment: banner.gif___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] hardened gentoo (selinux) asterisk problem
On Fri, 2004-12-31 at 11:59 +0100, christophe de coninck wrote: Hey, I've just installed hardened gentoo with selinux and emerged the selinux policy's for asterisk and emerged asterisk after it, now whenever i want to run asterisk i get: Dec 31 11:56:46 WARNING[4248]: manager.c:1474 init_manager: Unable to bind socket: Cannot assign requested address Note that that is manager.c, means it is trying to open the manager port. Had nothing to do with SIP. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] hardened gentoo (selinux) asterisk problem
Thanks for noting me, it was late yesterday and early this morning ;) just checked out manager.conf and there was the fault, thnx On Fri, 2004-12-31 at 12:32, Steven Critchfield wrote: On Fri, 2004-12-31 at 11:59 +0100, christophe de coninck wrote: Hey, I've just installed hardened gentoo with selinux and emerged the selinux policy's for asterisk and emerged asterisk after it, now whenever i want to run asterisk i get: Dec 31 11:56:46 WARNING[4248]: manager.c:1474 init_manager: Unable to bind socket: Cannot assign requested address Note that that is manager.c, means it is trying to open the manager port. Had nothing to do with SIP. -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] attachment: banner.gif___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP parameters
Sirs,
According to RFC 2705 (MGCP), these are the parameters that are used in the
transactions:
ReturnCode,
Connection-parameters
-- DeleteConnection(CallId,
EndpointId,
ConnectionId,
[Encapsulated NotificationRequest,]
[Encapsulated EndpointConfiguration])
ReturnCode,
-- DeleteConnection( CallId,
EndpointId,
ConnectionId,
Reason-code,
Connection-parameters)
ReturnCode,
-- DeleteConnection( CallId,
EndpointId)
ReturnCode,
-- DeleteConnection( EndpointId)
ReturnCode,
EndPointIdList|{
[RequestedEvents,]
[DigitMap,]
[SignalRequests,]
[RequestIdentifier,]
[NotifiedEntity,]
[ConnectionIdentifiers,]
[DetectEvents,]
[ObservedEvents,]
[EventStates,]
[BearerInformation,]
[RestartReason,]
[RestartDelay,]
[ReasonCode,]
[Capabilities]}
--- AuditEndPoint(EndpointId,
[RequestedInfo])
ReturnCode,
[CallId,]
[NotifiedEntity,]
[LocalConnectionOptions,]
[Mode,]
[RemoteConnectionDescriptor,]
[LocalConnectionDescriptor,]
[ConnectionParameters]
--- AuditConnection(EndpointId,
ConnectionId,
RequestedInfo)
ReturnCode,
[NotifiedEntity]
--- RestartInProgress ( EndPointId,
RestartMethod,
[RestartDelay,]
[Reason-code])
ReturnCode
-- EndpointConfiguration( EndpointId,
BearerInformation)
ReturnCode
-- NotificationRequest( EndpointId,
[NotifiedEntity,]
[RequestedEvents,]
RequestIdentifier,
[DigitMap,]
[SignalRequests,]
[QuarantineHandling,]
[DetectEvents,]
[encapsulated EndpointConfiguration])
ReturnCode
-- Notify( EndpointId,
[NotifiedEntity,]
RequestIdentifier,
ObservedEvents)
ReturnCode,
ConnectionId,
[SpecificEndPointId,]
[LocalConnectionDescriptor,]
[SecondEndPointId,]
[SecondConnectionId]
--- CreateConnection(CallId,
EndpointId,
[NotifiedEntity,]
[LocalConnectionOptions,]
Mode,
[{RemoteConnectionDescriptor |
SecondEndpointId}, ]
[Encapsulated NotificationRequest,]
[Encapsulated EndpointConfiguration])
ReturnCode,
[LocalConnectionDescriptor]
--- ModifyConnection(CallId,
EndpointId,
ConnectionId,
[NotifiedEntity,]
[LocalConnectionOptions,]
[Mode,]
[RemoteConnectionDescriptor,]
[Encapsulated NotificationRequest,]
[Encapsulated EndpointConfiguration])
My questions:
1) I am using Asterisk and a softphone called MGCP eyeP Phone. I was
watching the traffic between them at Ethereal and observed that some of them
have extra parameters. Example: CreateConnection has Request Identifier
(X), that is not described on RFC 2705. Should I ignore or consider it?
2) There are some parameters that don't have identifier? I mean, Request
Identifier is X:; Observed Events is O: ; Call ID is C: ... These
are them:
- Notified Entity
- Remote Connection Descriptor
- encapsulated Endpoint Configuration
- Second Endpoint Id
- encapsulated Notification Request
2.1) How can I identify them when they exist?
2.2) What means encapsulated parameters??
Thanks in advance,
Leonardo
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[Asterisk-Users] IAX users
Hi, I do not understand the difference between SIP and IAX, is it only two different protocols or something more special. The problem I have is that I've created two users Aix.conf register = users1:passwd1 register = user2:passwd2 [user1] type=user context=default secret=passwd1 host=dynamic [user2] type=user context=default secret=passwd2 host=dynamic extensions.conf exten = 550,1(Dial,IAX/user1); exten = 551,1(Dial,IAX/user2); and the error I get : Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No application 'IAX/user1)' for extension (default, 550, 1) == Spawn extension (default, 550, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]:1059/1' -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' Can someone help me how to get both users connected ? Thank you, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help
I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below and would appreciate any insights the experts can offer. Problem 1) The server is given its IP address using DHCP from my residential DSL gateway. The DNS settings are those from my ISP. Therefore the name allocated to the server (Vigor15) cannot be resolved to an IP address using DNS lookup. Other programs do not seem to be affected by this. I've fixed this by adding the name into the /etc/host.conf file, but wondered if this was an issue with the application (asterisk) or more generally my setup. I'm not sure if this is related to a problem where SIP, IAX protocols are set to listen on IP address 0.0.0.0 as in 2 below. Problem 2) SIP softPhones can't register. I think this may be due to listening on the wrong IP address 0.0.0.0:5060. Here's the log during startup: chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' Problem 3) Sometimes the program crashes during (at end of) the startup sequence. Warning about flexibel rate not heavily tested. Is this just a codec I can configure off/disable, or is this a crucial part of the system that will hopefully be fixed soon. I got this problem both with the latest stable release 1.0.1-2 (included in Mandrake) as well as the latest CVS-HEAD version checked out and rebuilt. The crash might be related to (4) below. cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) [EMAIL PROTECTED] david]# Warning, flexibel rate not heavily tested! Problem 4) Asterisk grabs the sound card for console use by default on startup. Its therefore not possible/easy to run KPhone or similar which also requires that resource. How can I turn off/stop asterisk trying to use the soundcard, and what are the implications. TIA Paidup System is P3 800MHz with 512MB ram, 19GB Disk, 100MBit Ethernet. Mandrake Linux Official 10.1. Similar problems with both recent stable release asterisk 1.0.1-2 and CVS_HEAD. SOftphone is KPhone (using SIP) on same machine. -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help
Might be related to the musiconhold files using different encoding rates ? Just an idea, also a newbie :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paid Up Sent: vendredi 31 décembre 2004 14:01 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below and would appreciate any insights the experts can offer. Problem 1) The server is given its IP address using DHCP from my residential DSL gateway. The DNS settings are those from my ISP. Therefore the name allocated to the server (Vigor15) cannot be resolved to an IP address using DNS lookup. Other programs do not seem to be affected by this. I've fixed this by adding the name into the /etc/host.conf file, but wondered if this was an issue with the application (asterisk) or more generally my setup. I'm not sure if this is related to a problem where SIP, IAX protocols are set to listen on IP address 0.0.0.0 as in 2 below. Problem 2) SIP softPhones can't register. I think this may be due to listening on the wrong IP address 0.0.0.0:5060. Here's the log during startup: chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' Problem 3) Sometimes the program crashes during (at end of) the startup sequence. Warning about flexibel rate not heavily tested. Is this just a codec I can configure off/disable, or is this a crucial part of the system that will hopefully be fixed soon. I got this problem both with the latest stable release 1.0.1-2 (included in Mandrake) as well as the latest CVS-HEAD version checked out and rebuilt. The crash might be related to (4) below. cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) [EMAIL PROTECTED] david]# Warning, flexibel rate not heavily tested! Problem 4) Asterisk grabs the sound card for console use by default on startup. Its therefore not possible/easy to run KPhone or similar which also requires that resource. How can I turn off/stop asterisk trying to use the soundcard, and what are the implications. TIA Paidup System is P3 800MHz with 512MB ram, 19GB Disk, 100MBit Ethernet. Mandrake Linux Official 10.1. Similar problems with both recent stable release asterisk 1.0.1-2 and CVS_HEAD. SOftphone is KPhone (using SIP) on same machine. -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help
Hi, 1) 0.0.0.0 just means listning on all interfaces and their ip adresses, not a problem. 2) Do a set verbose 100 to see if you have any communication with the sip phones or startup asterisk with asterisk -vvvggg 3) This is because a MPG3 file used for music on hold isn't support or that the Mandrake mpg123 is a wrong version 4) Try unloading the ALSA module in modules.conf Kind Regards Claus - Original Message - From: Paid Up [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 31, 2004 2:00 PM Subject: [Asterisk-Users] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below and would appreciate any insights the experts can offer. Problem 1) The server is given its IP address using DHCP from my residential DSL gateway. The DNS settings are those from my ISP. Therefore the name allocated to the server (Vigor15) cannot be resolved to an IP address using DNS lookup. Other programs do not seem to be affected by this. I've fixed this by adding the name into the /etc/host.conf file, but wondered if this was an issue with the application (asterisk) or more generally my setup. I'm not sure if this is related to a problem where SIP, IAX protocols are set to listen on IP address 0.0.0.0 as in 2 below. Problem 2) SIP softPhones can't register. I think this may be due to listening on the wrong IP address 0.0.0.0:5060. Here's the log during startup: chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' Problem 3) Sometimes the program crashes during (at end of) the startup sequence. Warning about flexibel rate not heavily tested. Is this just a codec I can configure off/disable, or is this a crucial part of the system that will hopefully be fixed soon. I got this problem both with the latest stable release 1.0.1-2 (included in Mandrake) as well as the latest CVS-HEAD version checked out and rebuilt. The crash might be related to (4) below. cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) [EMAIL PROTECTED] david]# Warning, flexibel rate not heavily tested! Problem 4) Asterisk grabs the sound card for console use by default on startup. Its therefore not possible/easy to run KPhone or similar which also requires that resource. How can I turn off/stop asterisk trying to use the soundcard, and what are the implications. TIA Paidup System is P3 800MHz with 512MB ram, 19GB Disk, 100MBit Ethernet. Mandrake Linux Official 10.1. Similar problems with both recent stable release asterisk 1.0.1-2 and CVS_HEAD. SOftphone is KPhone (using SIP) on same machine. -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Incoming Calls
On Tuesday 28 December 2004 04:32, C F wrote: Just a note on this. I tried using an external device with the TDM400 configured as 4 FXO to ring even with asterisk. But no matter how I configured it, asterisk always picked up. and the external device didn't ring (just the first ring for CallerID to come in). Asterisk should only pick up in 1 of 3 conditions: 1) you have an answer() statement in your dial plan 2) your dial plan dials the extensions and one of the IP phones picks up the call. 3) Asterisk drops the call into voicemail. I'm thinking that you probably have an answer() statement in your dial plan before the dial() statement. Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] FXO to IAX on ethernet. or FXO to SIP on Ethernet
Now I have searched around and not seen anything to do this. I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote locations like smaller towns so they would need to be simple, stable and require little to no user intervention after they are installed. Does anyone know of any hardware that will do this or a way that this could be done or ?? Sounds like you want something like the Sipura SPA-3000, which has one fxo port (pstn), one fxs port and one Ethernet (voip) port. About as small as it can get, remotely configurable via a browser, very stable, no buttons or screens, and a good selection of codecs for adjusting to small dsl bandwidth. Downside: more config options then you care to imagine; a little difficult to initially configure if you're not heavily into voip and telephony stuff. About $100 US. I've never played with a spa-3000 in a nat environment, but since several of the itsp's support it, it must work okay. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fault Problem
Hi, What do you think that the problem might be if a program has a segmentation fault at the same library call? The library call is from libpthread.so.0 and the call itself is pthread_mutex_locl ( ). I have enclosed the core dump information below. The program comes up and then does the segmentation fault. (gdb) bt #0 0x40035944 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 #1 0x419ede1e in register_verify (p=3D0x8163890, sin=3D0x41bfc0d0, req=3D0x41bfc0e0, uri=3D0x41bfc2fd sip:10.0.0.5, ignore=3D0) at chan_sip.c:5032 #2 0x419f4495 in handle_request (p=3D0x8163890, req=3D0x41bfc0e0, sin=3D0x41bfc0d0, recount=3D0x0, nounlock=3D0x41bfbf48) at chan_sip.c:7861 #3 0x419f6194 in sipsock_read (id=3D0x81413f8, fd=3D20, events=3D1, ignore=3D0x0) at chan_sip.c:7959 #4 0x0805317c in ast_io_wait (ioc=3D0x813fe08, howlong=3D0) at io.c:267 #5 0x419e27db in do_monitor (data=3D0x0) at chan_sip.c:8106 #6 0x400347f3 in start_thread () from /lib/tls/libpthread.so.0 #7 0x401a462a in clone () from /lib/tls/libc.so.6 (gdb) x/5i $eip 0x40035944 pthread_mutex_lock+36: mov0xc(%esi),%ecx 0x40035947 pthread_mutex_lock+39: cmp$0x1,%ecx 0x4003594a pthread_mutex_lock+42: je 0x40035974 pthread_mutex_lock+84 0x4003594c pthread_mutex_lock+44: jg 0x4003598e pthread_mutex_lock+110 0x4003594e pthread_mutex_lock+46: xor%eax,%eax (gdb) info registers eax0x0 0 ecx0x0 0 edx0x0 0 ebx0x4003dff4 1073995764 esp0x41bf01f4 0x41bf01f4 ebp0x41bf0208 0x41bf0208 esi0x0 0 edi0x32da 13018 eip0x40035944 0x40035944 eflags 0x210212 2163218 cs 0x73 115 ss 0x7b 123 ds 0x7b 123 es 0x7b 123 fs 0x0 0 gs 0x33 51 (gdb) info threads 15 process 13005 0xe410 in ?? () 14 process 13007 0xe410 in ?? () 13 process 13008 0xe410 in ?? () 12 process 13009 0xe410 in ?? () 11 process 13010 0xe410 in ?? () 10 process 13012 0xe410 in ?? () 9 process 13013 0xe410 in ?? () 8 process 13014 0xe410 in ?? () 7 process 13015 0xe410 in ?? () 6 process 13016 0xe410 in ?? () 5 process 13017 0xe410 in ?? () 4 process 13019 0xe410 in ?? () 3 process 13020 0xe410 in ?? () 2 process 13021 0xe410 in ?? () * 1 process 13018 0x40035944 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 asterisk:~ # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 10 cpu MHz : 996.859 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 1978.36 Thanks, Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FXO to IAX on ethernet. or FXO to SIP on Ethernet
Checkout http://www.mediatrix.com (FXO device 1204) or http://www.multitech.com. I have been looking into this myself. It appears that Nortel has an arrangement with Mediatrix and uses these devices where a remote FXO is needed that would be cost prohibitive to put in a full chassis. Avaya appears to have the same type of arrangement with Avaya where a G700 chassis is overkill. On both fronts I am *assuming* the quality and echo can is excellent if these two players are endorsing this solution. However, they are not in the price range of the products most of us have been using for FXO interfaces on this list. They may not also have the feature versatility we would like in a SOHO environment as their primary market will focus on quality but with dedicated purpose. The Mediatrix is a 4 port FXO only. MultiTech offer more units in different port counts, but each port appears to have flexible config options (FXO/FXS/EM, etc.) which adds significantly to the price. Mediatrix is list price 650.USD and the 2 port MultiTech looks to be 900. USD list. dbc. -- David Cook Quoting [EMAIL PROTECTED]: I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote locations like smaller towns so they would need to be simple, stable and require little to no user intervention after they are installed. Does anyone know of any hardware that will do this or a way that this could be done or ?? Thanks David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thanks for help - Almost done - 50% - Can hear
Hi, This is a thank you message for all that helped me including Max from www.asterisk-support.ru with whishes of a Happy New Year. Althought I still have a problem I'm happier I've 50% of my task complete. I'm using two TA from Draytek (router 2600V /router 2500V) 3 ADSL lines (2 for TA 1 for ASTERISK with a Draytek 2500 no voice model). If I call from one to another (using it's own mechnanism dialing #213*63*5*131#) I can connect and hear/speak both ends everything okay... If I register them on asterisk and call one another, the person I call can't hear me. Even registered with voipjet.com connected to them in IAX called my landline same happened. From the landline everything okay, from the voip phone i could hear but couldn't make myself heard. I'm guessing some funny port range left un-opened on the server router, but... I have at this point 5060 UDP and bot UDP and TCP from 5061 to 2. I have 1 to 2 in * configs for RTP. If anyone has an idea thanks in advance and a Happy New Year, Best regards from Portugal, Helder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IAX users
IAX2 - Original Message - From: Serge Schumacher [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, December 31, 2004 8:00 AM Subject: [Asterisk-Users] IAX users Hi, I do not understand the difference between SIP and IAX, is it only two different protocols or something more special. The problem I have is that I've created two users Aix.conf register = users1:passwd1 register = user2:passwd2 [user1] type=user context=default secret=passwd1 host=dynamic [user2] type=user context=default secret=passwd2 host=dynamic extensions.conf exten = 550,1(Dial,IAX/user1); exten = 551,1(Dial,IAX/user2); and the error I get : Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No application 'IAX/user1)' for extension (default, 550, 1) == Spawn extension (default, 550, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]:1059/1' -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' Can someone help me how to get both users connected ? Thank you, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] IAX users
Sorry ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: vendredi 31 décembre 2004 16:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX users IAX2 - Original Message - From: Serge Schumacher [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, December 31, 2004 8:00 AM Subject: [Asterisk-Users] IAX users Hi, I do not understand the difference between SIP and IAX, is it only two different protocols or something more special. The problem I have is that I've created two users Aix.conf register = users1:passwd1 register = user2:passwd2 [user1] type=user context=default secret=passwd1 host=dynamic [user2] type=user context=default secret=passwd2 host=dynamic extensions.conf exten = 550,1(Dial,IAX/user1); exten = 551,1(Dial,IAX/user2); and the error I get : Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No application 'IAX/user1)' for extension (default, 550, 1) == Spawn extension (default, 550, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]:1059/1' -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' Can someone help me how to get both users connected ? Thank you, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] IAX users
SIP is a XML-like control channel and is used to negotiate a separate RTP channel which carries the audio. It is complicated to set-up in cases of firewalls and NAT, but is an open standard. IAX2 is a candidate open standard and merges all traffic onto a single UDP stream - control and audio data. It has two modes, trunk and non-trunk. Trunk mode is highly efficient for transmitting multiple calls on a single UDP bearer and has minimal overhead. Standard IAX2 is easier to set-up than SIP. In terms of user experience, there should be little difference in call handling and audio quality - in general all of the same codecs and features are supported. IAX2 is a native protocol of Digium's Asterisk switch and I believe stands for Inter-Asterisk-eXchange version 2. To answer the query below, IAX (ie IAX11) was the precursor of IAX2. It is obsolete and should no longer be used. I use IAX when referring to IAX2, but obviously not all do! HTH Peter -Original Message- From: Serge Schumacher [mailto:[EMAIL PROTECTED] Sent: 31 December 2004 15:41 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX users Sorry ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: vendredi 31 décembre 2004 16:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX users IAX2 - Original Message - From: Serge Schumacher [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, December 31, 2004 8:00 AM Subject: [Asterisk-Users] IAX users Hi, I do not understand the difference between SIP and IAX, is it only two different protocols or something more special. The problem I have is that I've created two users Aix.conf register = users1:passwd1 register = user2:passwd2 [user1] type=user context=default secret=passwd1 host=dynamic [user2] type=user context=default secret=passwd2 host=dynamic extensions.conf exten = 550,1(Dial,IAX/user1); exten = 551,1(Dial,IAX/user2); and the error I get : Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No application 'IAX/user1)' for extension (default, 550, 1) == Spawn extension (default, 550, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]:1059/1' -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' Can someone help me how to get both users connected ? Thank you, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: FXO to IAX on ethernet. or FXO to SIP on Ethernet
Checkout http://www.mediatrix.com (FXO device 1204) or http://www.multitech.com. I have been looking into this myself. It appears that Nortel has an arrangement with Mediatrix and uses these devices where a remote FXO is needed that would be cost prohibitive to put in a full chassis. Avaya appears to have the same type of arrangement with Avaya where a G700 chassis is overkill. On both fronts I am *assuming* the quality and echo can is excellent if these two players are endorsing this solution. However, they are not in the price range of the products most of us have been using for FXO interfaces on this list. They may not also have the feature versatility we would like in a SOHO environment as their primary market will focus on quality but with dedicated purpose. The Mediatrix is a 4 port FXO only. MultiTech offer more units in different port counts, but each port appears to have flexible config options (FXO/FXS/EM, etc.) which adds significantly to the price. Mediatrix is list price 650.USD and the 2 port MultiTech looks to be 900. USD list. Unless Mediatrix has dramatically changed the 1204, be carefull with it. I tried to deploy one about nine months ago, and got it to work, but the config was very none standard even with the latest firmware. The 1204 did not have any form of 'register' support, no security (snmp is the only way to configure the box using the public community string and no way to change or protect it), and was no where near sip rfc compliant. It had excellent echo suppression, etc. However, given the changes that have been occuring with asterisk code, there is a very high probability interaction with the 1204 would fail, and Mediatrix offers no upgrade support other then 'pay as you go' for each firmware release. Support is only offered through their resellers, and the majority of those are traditional pbx dealers that start with what is asterisk? The 1204 was specifically designed to interoperate with the 1104 as a toll-bypass combination that happened to use sip. If Mediatrix would take rfc compliance seriously, it would make a very nice 4-port fxo, although it is still a little pricey. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] RealTime Drivers Connectivity Error
Use res_config_odbc ... Lots of people have problems with the mysql one. I have never once had a problem with the odbc one. The wiki even has a small how to on setting it up. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of adnan Sent: Thursday, December 30, 2004 2:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RealTime Drivers Connectivity Error Hello *'s, i am using Realtime Sip drivers but its not working here is my configs: extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends = mysql,asterisk,sip_friends res_mysql.conf [general] dbhost = localhost.localdoamin/127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = 123456 dbport = 3306 dbsock = /var/lib/mysql/mysql.sock error detail: Dec 31 01:20:49 ERROR[4298]: res_config_mysql.c:617 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on localhost.localdomain/127.0.0.1. Check debug for more info. == Registered application 'UserEvent' [app_verbose.so]Segmentation fault (core dumped) i change dbhost parameter several times like(localhost,192.168.10.193 etc) but can't works I am using latest CVS-Head kindly pointout my mistakes. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FXO to IAX on ethernet. or FXO to SIP on Ethernet
There are two different approaches available: 1. Hardware What you want is a remote SIP gateway. These are boxes which have FXS/FXO/EM ports in some combination on one side and and an ethernet port on the other. Most of these boxes were originally designed to run H.323 and had SIP firmware added at a later stage. An example is http://www.ovislink.com.tw/voip400.htm. Ovislink have 4 and 8 port units. Each takes a 4-port (or two 4-port units for the 8 port model) adapters. These can be either four FXS or four FXO or four EM adapters. Ovislink also have a 2-port model http://www.ovislink.com.tw/voip220rs.htm which seems to have better SIP support. These boxes were originally designed to run H.323. Since SIP has become popular Ovislink added a SIP frontend component to their firmware. This has the effect that you have to configure *both* the H.323 component *and* the SIP component to get the box going. It does work, but it can be a pain to configure (try reading the fairly comprhensive manual two or three time and then having two or three goes at it before it all works). Read the user manual (http://www.ovislinkcorp.com/Manuals/VoIP800-400%20manual.pdf), the separate SIP guide (http://www.ovislinkcorp.com/Manuals/SIP_Guide.pdf) and the VOIP command reference (http://www.ovislinkcorp.com/Manuals/VoIPReference.pdf) to figure out whether they do what you want. These boxes can be rediculously cheap on occasion. I've seen new Ovislink 8-port gateways on eBay for US$200-300 form time to time. Otherwise, I believe that they have distributors in the US/Europe/Australia. The main problem with using the Ovislink gateways is making sure that they have the correct approvals. For instance I found that I couldn't use one here in Australia because they lack A-tick approval (and I'm not about to spend the $50K needed to get them tested). They *appear* to have FCC and CE approval, but they would not be the first manufacturer to print approval numbers on the case when the approvals did not actually exist. I'd check before I'd use one - using non type-approved equipment can attract very large fines. In general, these boxes are reasonably reliable, or at least reliable as say an ADSL modem/router. If the location was really remote you could place a second box at the loaction and a PSTN switch to switch the lines. Hopefully there would be someone on the premises who could unplug the PSTN line from box-A and connect them to box-B if necessary. 2. Telco/Service Provider I don't use the Ovislink box myself, although I did evaluate them. After I hit the lack of approvals roadblock I mention above, I took a very different and much simpler approach. I found a telco who would do call collection for me. They had Cisco routers in each telephone district in Australia. Incoming calls on my numbers were sent to their routers which sent them directly to my gateway. Now admittedly this was for a much larger application than you are talking about (60 lines - actually telephone numbers) are involved. The biggest problem was that the telco would only deliver the calls using H.323 (since most business PABX's use H.323 rather than SIP), so I had to build a H.32-to-SIP gateway using asterisk (which was a pain to get going - asterisk's H.323 support is ideosyncratic). On one hand the telco approach was cheaper (a monthly charge rather than having to buy and house a number of routers). On the other hand it is an ongoing charge. From memory, the hardware cost represented about 30-40 months of telco charges. The compelling reasons for choosing the telco approach are (a) simplicity - its a lot simpler to have one gateway rather than a number of different PSTN gateways in remote locations; (b) reliability - the telco has around $175M is Cisco kit, if something breaks they have a redundent backup standing by. I hope this gives you a few pointers. regards Matthew Date: Fri, 31 Dec 2004 18:22:03 +1100 From: David Uzzell [EMAIL PROTECTED] Subject: [Asterisk-Users] FXO to IAX on ethernet. or FXO to SIP on Ethernet To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Now I have searched around and not seen anything to do this. I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote locations like smaller towns so they would need to be simple, stable and require little to no user intervention after they are installed. Does anyone know of any hardware that will do this or a way that this could be done or ?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To
[Asterisk-Users] BroadVoice WiSIP with Asterisk
BroadVoice sells a wireless SIP phone for $149. Does this phone as sold by BroadVoice work with Asterisk or is it a locked down device like the Vonages ATA186? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IAX users
On Fri, 31 Dec 2004, Serge Schumacher wrote: extensions.conf exten = 550,1(Dial,IAX/user1); exten = 551,1(Dial,IAX/user2); and the error I get : Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No application 'IAX/user1)' for extension (default, 550, 1) == Spawn extension (default, 550, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]:1059/1' -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' Can someone help me how to get both users connected ? You've got two issues. 1) You need to say IAX2, not IAX. (the IAX in your Asterisk is actually the version 2) 2) Your extensions.conf syntax is messed up. So they should look like: exten = 550,1,Dial(IAX2/user1) exten = 551,1,Dial(IAX2/user2) Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broken pipe...
Hello, I've done a very straightforward install of Asterisk, and can't seem to get it started. This is a proof-of-concept installation, and currently does not have any T1/E1 or FXO/FXS cards in it. I just want to use it as an internal SIP server for now. However, when I try to start Asterisk, it dies with the following messages: Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I have googled for info on these errors - some have reported getting them while _exiting_ Asterisk, but none have claimed to see them while _starting_ Asterisk. Any ideas? Thanks a ton. --sk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Broken pipe...
Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe These are messages from mpg123 NOT Asterisk. bkw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] MGCP parameters
The RFC specification alone is not sufficient, there are many signaling
packages that are defined elsewhere.
Also, RFC 2705 is out of date, see RFC 3435
Leonardo J. Tramontina wrote:
Sirs,
According to RFC 2705 (MGCP), these are the parameters that are used
in the transactions:
ReturnCode,
Connection-parameters
-- DeleteConnection(CallId,
EndpointId,
ConnectionId,
[Encapsulated NotificationRequest,]
[Encapsulated EndpointConfiguration])
ReturnCode,
-- DeleteConnection( CallId,
EndpointId,
ConnectionId,
Reason-code,
Connection-parameters)
ReturnCode,
-- DeleteConnection( CallId,
EndpointId)
ReturnCode,
-- DeleteConnection( EndpointId)
ReturnCode,
EndPointIdList|{
[RequestedEvents,]
[DigitMap,]
[SignalRequests,]
[RequestIdentifier,]
[NotifiedEntity,]
[ConnectionIdentifiers,]
[DetectEvents,]
[ObservedEvents,]
[EventStates,]
[BearerInformation,]
[RestartReason,]
[RestartDelay,]
[ReasonCode,]
[Capabilities]}
--- AuditEndPoint(EndpointId,
[RequestedInfo])
ReturnCode,
[CallId,]
[NotifiedEntity,]
[LocalConnectionOptions,]
[Mode,]
[RemoteConnectionDescriptor,]
[LocalConnectionDescriptor,]
[ConnectionParameters]
--- AuditConnection(EndpointId,
ConnectionId,
RequestedInfo)
ReturnCode,
[NotifiedEntity]
--- RestartInProgress ( EndPointId,
RestartMethod,
[RestartDelay,]
[Reason-code])
ReturnCode
-- EndpointConfiguration( EndpointId,
BearerInformation)
ReturnCode
-- NotificationRequest( EndpointId,
[NotifiedEntity,]
[RequestedEvents,]
RequestIdentifier,
[DigitMap,]
[SignalRequests,]
[QuarantineHandling,]
[DetectEvents,]
[encapsulated EndpointConfiguration])
ReturnCode
-- Notify( EndpointId,
[NotifiedEntity,]
RequestIdentifier,
ObservedEvents)
ReturnCode,
ConnectionId,
[SpecificEndPointId,]
[LocalConnectionDescriptor,]
[SecondEndPointId,]
[SecondConnectionId]
--- CreateConnection(CallId,
EndpointId,
[NotifiedEntity,]
[LocalConnectionOptions,]
Mode,
[{RemoteConnectionDescriptor |
SecondEndpointId}, ]
[Encapsulated NotificationRequest,]
[Encapsulated EndpointConfiguration])
ReturnCode,
[LocalConnectionDescriptor]
--- ModifyConnection(CallId,
EndpointId,
ConnectionId,
[NotifiedEntity,]
[LocalConnectionOptions,]
[Mode,]
[RemoteConnectionDescriptor,]
[Encapsulated NotificationRequest,]
[Encapsulated EndpointConfiguration])
My questions:
1) I am using Asterisk and a softphone called MGCP eyeP Phone. I was
watching the traffic between them at Ethereal and observed that some
of them have extra parameters. Example: CreateConnection has Request
Identifier (X), that is not described on RFC 2705. Should I ignore or
consider it?
2) There are some parameters that don't have identifier? I mean,
Request Identifier is X:; Observed Events is O: ; Call ID is
C: ... These are them:
- Notified Entity
- Remote Connection Descriptor
- encapsulated Endpoint Configuration
- Second Endpoint Id
- encapsulated Notification Request
2.1) How can I identify them when they exist?
2.2) What means encapsulated
Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk
On Fri, 31 Dec 2004, Adi Linden wrote: BroadVoice sells a wireless SIP phone for $149. Does this phone as sold by BroadVoice work with Asterisk or is it a locked down device like the Vonages ATA186? You'd probably have to ask them that. Just so you know, you can buy that phone elsewhere. It is made by Pulver Innovations (www.pulverinnovations.com). The fact that it lists at $199 on Pulver's site suggests that it would probably be tethered to the BroadVoice service, or at the very least you can count on paying the difference in the disconnect fee whenever you close your account at BroadVoice (they don't chage a disconnect fee if you brought your own device instead of buying a discounted model from them). Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] BroadVoice WiSIP with Asterisk
BroadVoice sells a wireless SIP phone for $149. Does this phone as sold by BroadVoice work with Asterisk or is it a locked down device like the Vonages ATA186? You'd probably have to ask them that. Just so you know, you can buy that phone elsewhere. It is made by Pulver Innovations (www.pulverinnovations.com). The fact that it lists at $199 on Pulver's site suggests that it would probably be tethered to the BroadVoice service, or at the very least you can count on paying the difference in the disconnect fee whenever you close your account at BroadVoice (they don't chage a disconnect fee if you brought your own device instead of buying a discounted model from them). I didn't look into any disconnect fees yet. That's a good one to be aware of since the phone appears to be available in a bundle with their service. I had a look at the Pulver cordless. How does it (at $199) compare to the Zyxel 2000W (~$250 from voipsupply.com)? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] BroadVoice WiSIP with Asterisk
I didn't look into any disconnect fees yet. That's a good one to be aware of since the phone appears to be available in a bundle with their service. I had a look at the Pulver cordless. How does it (at $199) compare to the Zyxel 2000W (~$250 from voipsupply.com)? Adi They are the same phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Final call for departments
Thanks to your generous donations, people all over will continue to hear such great programming as Fresh Air, All Things Considered, Morning Edition and the BBC World News... Sorry, I meant to say - such great programming as 'Barn', 'Food Services', 'Security' and 'I.T. Services' Thats correct, we made our goal! We should have the attached list back in a few weeks (after they are recorded and Rob edits them). As for those from bug 3006 I am not sure. I believe Rob was including them (I was just putting together a list of departments.), but I will check with him to be sure. I would really like to thank everyone for helping out to make Asterisk what it is. James Alspach Family wrote: Today is the day. The most up to date list is attached. I will forward it to Rob Friday night so, anything you want and can get to be by then I will add. Do the phrases being send include those in bug 3006? If not, can they? Lets say half of them are active readers and half of those are using Asterisk seriously. If those 250 people remaining could just donate $.50 each, we would reach our goal.. Public television and radio stations everywhere would be proud of you. Nick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users asterisk departments.sxc Description: OpenOffice Calc spreadsheet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura 3000 inbound FXO problem
you have two 'friend' entries in your sip.conf. it uses the second, which is not what you want. one should be peer and the other user. though a number of versions of asterisk don't actually work with peer/user, a major pita. so try reversing the order of the two entries if you have problems with peer/user. and don't post a bug report about this. you will get screamed at and insulted (seems the mentality of the asterisk community), and told you should have posted your question to this list despite your already having done so. this kind of response/support is why we went with a commercial solution for production; though i keep * for home use. randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] job offer
A friend of mine urgently needs a skilled asterisk, linux, Mysql type person for a one off job. Preferrably London based. Email me with your details if your interested. Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX media
Hi ALL; In IAX protocol, both rtp and signaling are handled on the same port, so the Asterisk is always in the path of rtp traffic. Am I right? If yes, is there anyway to set Asterisk just as signal proxy ? Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rdnis
Anyone have a example of how to setup RDNIS in *? To date we have been giving each voicemail user a individual DNIS but would like to consolidate all the numbers into one and just use RDNIS to route the call. Thanks, Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manager API / weird queue
Hi,
I'm playing with the agent/queue system. Everything work well with v1.0.3.
but I want the 'Action: Agents' in the manager API that is only on the CVS
version. So i switched to, but now the Queue/Agent system barely work. (my
agent don't get the call)
Where I can get a 'stable' CVS version?
Or maybe, how I can solve my Queue/Problem? here is the detail:
1. I can login as an Agent with that extension:
[cc-queues]
; nos queues
exten = s,1,Answer
exten = s,2,Queue(queue1)
exten = s,3,Hangup
asterisk console log:
-- Executing AgentLogin(SIP/samuel-b5fd, 11) in new stack
Urgent handler
-- Playing 'agent-pass' (language 'en')
Urgent handler
-- Playing 'agent-loginok' (language 'en')
Urgent handler
-- Started music on hold, class 'default', on SIP/samuel-b5fd
Urgent handler
== Agent '11' logged in (format ulaw/slin)
Urgent handler
2. Next step, I generate a call in the spool/outgoing
cp anwser.call /usr/asterisk/var/spool/asterisk/outgoing/rwerewewr
content of my call file:
Channel: IAX2/[EMAIL PROTECTED]:5036/1555252532442422
MaxRetries: 0
Application: Queue
Data: queue1
note: that dial an extension with the SayDigits(${EXTEN}) application.
(just to hear something)
asterisk console output:
-- Attempting call on IAX2/[EMAIL PROTECTED]:5036/1555252532442422 for
application Queue(queue1) (Retry 1)
Urgent handler
Urgent handler
Urgent handler
-- Accepting unauthenticated call from 127.0.0.1, requested format =
64, actual format = 64
Urgent handler
-- Call accepted by 127.0.0.1 (format slin)
-- Format for call is slin
Urgent handler
-- Executing Answer(IAX2/[EMAIL PROTECTED]:5036/3, ) in new stack
Urgent handler
Urgent handler
Dec 31 16:20:24 WARNING[22251]: app_queue.c:2094 queue_exec: Unable to
join queue 'queue1'
Urgent handler
Dec 31 16:20:24 NOTICE[22218]: chan_iax2.c:1391 iax2_destroy: Avoiding IAX
destroy deadlock
-- Hungup 'IAX2/127.0.0.1:5036/1'
Urgent handler
-- Executing SayDigits(IAX2/[EMAIL PROTECTED]:5036/3,
555252532442422) in new stack
Urgent handler
-- Playing 'digits/5' (language 'en')
Urgent handler
-- Hungup 'IAX2/[EMAIL PROTECTED]:5036/3'
Urgent handler
Dec 31 16:20:24 NOTICE[22251]: pbx_spool.c:244 attempt_thread: Call
completed to IAX2/[EMAIL PROTECTED]:5036/1555252532442422
thanks!
Samuel T. Cossette
[EMAIL PROTECTED], 1.418.8o2.784o
Well, that's for me to know and you to find out. Jeffrey, Blue Velvet
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[Asterisk-Users] Mysql-Realtime and ASTCC
Hi ALL; Hi Matthew; Can we integerate ASTCC Sipfriend or Iaxfrien tables with Mysql-Realtime driver? Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX users
Whisker, Peter [EMAIL PROTECTED] writes: SIP is a XML-like control channel [...] I think you meant HTTP-like. :-) -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mysql-Realtime and ASTCC
If you look through the astcc-admin source a little, you should find it quite easy to modify astcc to use the realtime drivers. If you get it changed, please submit it as a bug report. That is something that needs to get changed. Darren Wiebe [EMAIL PROTECTED] mohammad wrote: Hi ALL; Hi Matthew; Can we integerate ASTCC Sipfriend or Iaxfrien tables with Mysql-Realtime driver? Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ????
double post ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Final call for departments
How about Technical Services? On Fri, 2004-12-31 at 13:32, Alspach Family wrote: Thanks to your generous donations, people all over will continue to hear such great programming as Fresh Air, All Things Considered, Morning Edition and the BBC World News... Sorry, I meant to say - such great programming as 'Barn', 'Food Services', 'Security' and 'I.T. Services' Thats correct, we made our goal! We should have the attached list back in a few weeks (after they are recorded and Rob edits them). As for those from bug 3006 I am not sure. I believe Rob was including them (I was just putting together a list of departments.), but I will check with him to be sure. I would really like to thank everyone for helping out to make Asterisk what it is. James Alspach Family wrote: Today is the day. The most up to date list is attached. I will forward it to Rob Friday night so, anything you want and can get to be by then I will add. Do the phrases being send include those in bug 3006? If not, can they? Lets say half of them are active readers and half of those are using Asterisk seriously. If those 250 people remaining could just donate $.50 each, we would reach our goal.. Public television and radio stations everywhere would be proud of you. Nick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Softphone in German
I am looking for a German language softphone. Is there such a thing? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly lockup in Win98
I tried Firefly in WinXP and it works fine. I tried it on Win98, it looks it up. Anyone experience this? Any ideas? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 12/30/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FC3 compile with new 2.6.10 fails
All, I have FC3 fedora core 3 and just installed and compiled 2.6.10. after rebooting I attempted to recompile zaptel-1.0.3. I did a make clean then make. I got the following errors. Any suggestions? --- /usr/src/digium/zaptel-1.0.3/torisa.c:1139: warning: 'set_tor_base' defined but not used CC [M] /usr/src/digium/zaptel-1.0.3/wcusb.o CC [M] /usr/src/digium/zaptel-1.0.3/wcfxo.o CC [M] /usr/src/digium/zaptel-1.0.3/wcfxs.o /usr/src/digium/zaptel-1.0.3/wcfxs.c: In function `wcfxs_interrupt': /usr/src/digium/zaptel-1.0.3/wcfxs.c:473: sorry, unimplemented: inlining failed in call to 'wcfxs_proslic_check_hook': function body not available /usr/src/digium/zaptel-1.0.3/wcfxs.c:810: sorry, unimplemented: called from here /usr/src/digium/zaptel-1.0.3/wcfxs.c:474: sorry, unimplemented: inlining failed in call to 'wcfxs_proslic_recheck_sanity': function body not available /usr/src/digium/zaptel-1.0.3/wcfxs.c:812: sorry, unimplemented: called from here /usr/src/digium/zaptel-1.0.3/wcfxs.c:472: sorry, unimplemented: inlining failed in call to 'wcfxs_voicedaa_check_hook': function body not available /usr/src/digium/zaptel-1.0.3/wcfxs.c:814: sorry, unimplemented: called from here make[2]: *** [/usr/src/digium/zaptel-1.0.3/wcfxs.o] Error 1 make[1]: *** [_module_/usr/src/digium/zaptel-1.0.3] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.10' make: *** [linux26] Error 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help With Configuration From Odbc
Hi. I can't figure this one out. Hope someone can help me. [EMAIL PROTECTED]:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName= Password= Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings= [EMAIL PROTECTED]:# cat /etc/odbcinst.ini [PostgreSQL] Description=PostgreSQL ODBC driver for Linux and Windows Driver=/usr/local/lib/psqlodbc.so Setup=/usr/lib/odbc/libodbcpsqlS.so Debug = 1 CommLog = 1 [EMAIL PROTECTED]:# echo select * from ast_config where filename='iax.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id | isql Asterisk lot of output from table SQLRowCount returns 39 39 rows fetched So the odbc thingy works! [EMAIL PROTECTED]:# cat res_config_odbc.conf [settings] table = ast_config connection = myconn [EMAIL PROTECTED]:# cat res_odbc.conf [myconn] dsn=Asterisk username=X password=X preconnect=yes [EMAIL PROTECTED]:# cat extconfig.conf [settings] agents.conf = odbc enum.conf = odbc extensions.conf = odbc iax.conf = odbc iaxprov.conf = odbc queues.conf = odbc sip.conf = odbc zapata.conf = odbc And asterisk answers: [res_odbc.so] = (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:133 load_odbc_config: registered database handle 'myconn' dsn-[Asterisk] Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:379 load_module: res_odbc loaded. [res_config_odbc.so] = (ODBC Configuration) Jan 1 02:21:11 NOTICE[32024]: config.c:888 ast_config_register: Registered Config Engine odbc == Parsing '/etc/asterisk/extconfig.conf': Found Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: agents.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: enum.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: extensions.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iax.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iaxprov.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: queues.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: sip.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: zapata.conf to odbc Jan 1 02:21:11 NOTICE[32024]: res_config_odbc.c:190 load_module: res_config_odbc loaded. [skipping res_adsi.so] [skipping chan_modem.so] [chan_sip.so] = (Session Initiation Protocol (SIP)) Jan 1 02:21:11 NOTICE[32024]: config.c:764 __ast_load: Loading Config sip.conf via odbc engine == Parsing '/etc/asterisk/res_config_odbc.conf': Found Jan 1 02:21:11 WARNING[32024]: res_config_odbc.c:103 config_odbc: SQL select error! [select * from ast_config where filename='sip.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id] What is wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Help With Configuration From Odbc
Sorry about this. Just figured it out. In res_odbc.conf its supposed to be pre-connect and not preconnect. On Saturday 01 January 2005 02:30, Arthur B Olsen wrote: Hi. I can't figure this one out. Hope someone can help me. [EMAIL PROTECTED]:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName= Password= Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings= [EMAIL PROTECTED]:# cat /etc/odbcinst.ini [PostgreSQL] Description=PostgreSQL ODBC driver for Linux and Windows Driver=/usr/local/lib/psqlodbc.so Setup=/usr/lib/odbc/libodbcpsqlS.so Debug = 1 CommLog = 1 [EMAIL PROTECTED]:# echo select * from ast_config where filename='iax.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id | isql Asterisk lot of output from table SQLRowCount returns 39 39 rows fetched So the odbc thingy works! [EMAIL PROTECTED]:# cat res_config_odbc.conf [settings] table = ast_config connection = myconn [EMAIL PROTECTED]:# cat res_odbc.conf [myconn] dsn=Asterisk username=X password=X preconnect=yes [EMAIL PROTECTED]:# cat extconfig.conf [settings] agents.conf = odbc enum.conf = odbc extensions.conf = odbc iax.conf = odbc iaxprov.conf = odbc queues.conf = odbc sip.conf = odbc zapata.conf = odbc And asterisk answers: [res_odbc.so] = (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:133 load_odbc_config: registered database handle 'myconn' dsn-[Asterisk] Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:379 load_module: res_odbc loaded. [res_config_odbc.so] = (ODBC Configuration) Jan 1 02:21:11 NOTICE[32024]: config.c:888 ast_config_register: Registered Config Engine odbc == Parsing '/etc/asterisk/extconfig.conf': Found Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: agents.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: enum.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: extensions.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iax.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iaxprov.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: queues.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: sip.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: zapata.conf to odbc Jan 1 02:21:11 NOTICE[32024]: res_config_odbc.c:190 load_module: res_config_odbc loaded. [skipping res_adsi.so] [skipping chan_modem.so] [chan_sip.so] = (Session Initiation Protocol (SIP)) Jan 1 02:21:11 NOTICE[32024]: config.c:764 __ast_load: Loading Config sip.conf via odbc engine == Parsing '/etc/asterisk/res_config_odbc.conf': Found Jan 1 02:21:11 WARNING[32024]: res_config_odbc.c:103 config_odbc: SQL select error! [select * from ast_config where filename='sip.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id] What is wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how is a upgrade performed?
I have a stable server and want to upgrade. How do I upgrade to the latest version of * ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FC2 ztcfg - cannot find channel 2
When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) My /etc/zaptel.conf is: fxsks=1 fxoks=2 loadzone = au defaultzone=au Channel 1 is a X101P card connected to the PSTN and channel 2 is a S100U box driving an analogue phone. The zaptel kernel module gets loaded OK as does the wcfxs module for the X101P card and the wcusb module for the USB device but then I get the error message. In context the error message is in the following string when i run /etc/init.d/zaptel restart: Jan 1 10:48:16 bu kernel: usbcore: deregistering driver wcusb Jan 1 10:48:16 bu kernel: Freed a Wildcard Jan 1 10:48:16 bu kernel: Zapata Telephony Interface Unloaded Jan 1 10:48:16 bu udev[27206]: removing device node '/udev/zap1' Jan 1 10:48:17 bu udev[27207]: removing device node '/udev/zaptimer' Jan 1 10:48:17 bu udev[27222]: removing device node '/udev/zapchannel' Jan 1 10:48:17 bu udev[27233]: removing device node '/udev/zappseudo' Jan 1 10:48:17 bu udev[27244]: removing device node '/udev/zapctl' Jan 1 10:48:17 bu zaptel: Removing zaptel module: succeeded Jan 1 10:48:18 bu kernel: Zapata Telephony Interface Registered on major 196 Jan 1 10:48:18 bu zaptel: Loading zaptel framework: succeeded Jan 1 10:48:18 bu kernel: PCI: Found IRQ 11 for device :00:0b.0 Jan 1 10:48:18 bu udev[27287]: creating device node '/udev/zapctl' Jan 1 10:48:18 bu kernel: wcfxo: DAA mode is 'FCC' Jan 1 10:48:18 bu kernel: Found a Wildcard FXO: Wildcard X101P Jan 1 10:48:18 bu kernel: usbcore: registered new driver wcusb Jan 1 10:48:18 bu kernel: Wildcard USB FXS Interface driver registered Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) Jan 1 10:48:18 bu zaptel: Running ztcfg: failed Jan 1 10:48:18 bu udev[27266]: creating device node '/udev/zaptimer' Jan 1 10:48:19 bu udev[27273]: creating device node '/udev/zapchannel' Jan 1 10:48:19 bu udev[27280]: creating device node '/udev/zappseudo' Jan 1 10:48:19 bu udev[27300]: creating device node '/udev/zap1' I know that the logs refer to /udev but I have opted to create the /dev/zap nodes instead and have the following nodes: # ll /dev/zap/ total 0 crw--- 1 root root 196, 1 Jan 1 10:34 1 crw--- 1 root root 196, 2 Jan 1 10:37 2 crw--- 1 root root 196, 254 Jan 1 10:34 channel crw--- 1 root root 196, 0 Jan 1 10:34 ctl crw--- 1 root root 196, 255 Jan 1 10:34 pseudo crw--- 1 root root 196, 253 Jan 1 10:34 timer I am assuming here that /dev/zap/1 refers to channel 1 and /dev/zap/2 refers to channel 2 but that could be a wrong assumption. Any/all help would be appreciated. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RFC3389 support incomplete
We try a X-Lite client from remote to connect to my * I can call X-Lite and X-Lite can call me. However, X-Lite can hear my voice, while I cannot hear him. *CLI shows *CLI (date) NOTICE[4637] rtp.c:317 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible RFC3389: 5 bytes, level 4 ... I tried in my sip.conf to change dtmfmode from inband to info and to rfc2833 without success. Can anybody give me a hint? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fault (core dumped)
i am facing unusual and wiered error in asterisk using Realtime MYSQL driver . Asterisk runs well and smoothly with absoulutely no error or warning but everytime i power-on my sip-phone ,booting, initializes and then asterisk suddenly quit with the error. _*Segmentation Fault (core dumped)*_ i see in /var/log/messages,/var/log/asterisk/messages but all is clear no sign of any error message or warning, what does its mean its my configs problem or something wrong with asterisk i use Latest CVS. Can i use Realtime odbc instead of Mysql . extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends = mysql,asterisk,sip_friends res_mysql.conf [general] dbhost = localhost.localdoamin/127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = 123456 dbport = 3306 dbsock = /var/lib/mysql/mysql.sock ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
