[asterisk-users] calls dropped with te110p E1
I have this problem on a te110p. On random basis the calls are disconnected by asterisk. When this happens asterisk logs: Jul 18 12:52:10 DEBUG[4668] channel.c: Got a FRAME_CONTROL (5) frame on channel Zap/13-1 Jul 18 12:52:10 DEBUG[4668] channel.c: Bridge stops bridging channels SIP/201-08270aa8 and Zap/13-1 Please see the atteched files containing details. SETUP SETUP ACK ALLERTING CONNECT CONNECT ACK DISCONNECT RELEASE RELEASE ACK Asterisk 1.2.22 Zaptel 1.2.19 Libpri 1.2.5 Addons 1.2.7 Sounds 1.2.1 Tommaso Calosi Jul 18 12:51:48 DEBUG[4200] chan_sip.c: Checking SIP call limits for device 201 Jul 18 12:51:48 DEBUG[4200] chan_sip.c: build_route: Contact hop: Marzia Ridolfi sip:[EMAIL PROTECTED]:5060 Jul 18 12:51:48 VERBOSE[4187] logger.c: Extension Changed 201 new state InUse for Notify User 200 Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Executing NoOp(SIP/201-08270aa8, Real Number: 03356307910) in new stack Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Executing Macro(SIP/201-08270aa8, superdial|1|ZAP/G10|3356307910||tT) in new stack Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Executing NoOp(SIP/201-08270aa8, Peer 1 using ZAP/G10 to call 3356307910 for with options tT) in new stack Jul 18 12:51:48 DEBUG[4668] app_macro.c: Executed application: Noop Jul 18 12:51:48 DEBUG[4668] pbx.c: Function result is '201' Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Executing Set(SIP/201-08270aa8, CDR(accountcode)=0557870201) in new stack Jul 18 12:51:48 DEBUG[4668] app_macro.c: Executed application: Set Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Executing Set(SIP/201-08270aa8, CALLERID(all)=05578701 05578701) in new stack Jul 18 12:51:48 DEBUG[4668] app_macro.c: Executed application: Set Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Executing Dial(SIP/201-08270aa8, ZAP/G10/3356307910||tT) in new stack Jul 18 12:51:48 DEBUG[4668] dsp.c: dsp busy pattern set to 0,0 Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Making new call for cr 32791 Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Requested transfer capability: 0x00 - SPEECH Jul 18 12:51:48 VERBOSE[4668] logger.c: Protocol Discriminator: Q.931 (8) len=50 Jul 18 12:51:48 VERBOSE[4668] logger.c: Call Ref: len= 2 (reference 23/0x17) (Originator) Jul 18 12:51:48 VERBOSE[4668] logger.c: Message type: SETUP (5) Jul 18 12:51:48 VERBOSE[4668] logger.c: [04 03 80 90 a3] Jul 18 12:51:48 VERBOSE[4668] logger.c: Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Jul 18 12:51:48 VERBOSE[4668] logger.c: Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Jul 18 12:51:48 VERBOSE[4668] logger.c: Ext: 1 User information layer 1: A-Law (35) Jul 18 12:51:48 VERBOSE[4668] logger.c: [18 03 a9 83 8d] Jul 18 12:51:48 VERBOSE[4668] logger.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 Jul 18 12:51:48 VERBOSE[4668] logger.c: ChanSel: Reserved Jul 18 12:51:48 VERBOSE[4668] logger.c:Ext: 1 Coding: 0 Number Specified Channel Type: 3 Jul 18 12:51:48 VERBOSE[4668] logger.c:Ext: 1 Channel: 13 ] Jul 18 12:51:48 VERBOSE[4668] logger.c: [28 08 30 35 35 37 38 37 30 31] Jul 18 12:51:48 VERBOSE[4668] logger.c: Display (len= 8) [ 05578701 ] Jul 18 12:51:48 VERBOSE[4668] logger.c: [6c 0a 00 80 30 35 35 37 38 37 30 31] Jul 18 12:51:48 VERBOSE[4668] logger.c: Calling Number (len=12) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Jul 18 12:51:48 VERBOSE[4668] logger.c: Presentation: Presentation permitted, user number not screened (0) '05578701' ] Jul 18 12:51:48 VERBOSE[4668] logger.c: [70 0b 80 33 33 35 36 33 30 37 39 31 30] Jul 18 12:51:48 VERBOSE[4668] logger.c: Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '3356307910' ] Jul 18 12:51:48 DEBUG[4187] channel.c: Avoiding initial deadlock for 'Zap/13-1' Jul 18 12:51:48 VERBOSE[4668] logger.c: -- Called G10/3356307910 Jul 18 12:51:48 DEBUG[4200] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 114: Match Found Jul 18 12:51:48 DEBUG[4668] chan_sip.c: Oooh, format changed to 4 Jul 18 12:51:48 VERBOSE[4193] logger.c: Protocol Discriminator: Q.931 (8) len=10 Jul 18 12:51:48 VERBOSE[4193] logger.c: Call Ref: len= 2 (reference 23/0x17) (Terminator) Jul 18 12:51:48 VERBOSE[4193] logger.c: Message type: SETUP ACKNOWLEDGE (13) Jul 18 12:51:48 VERBOSE[4193] logger.c: [18 03 a9 83 8d] Jul 18 12:51:48 VERBOSE[4193] logger.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 Jul 18 12:51:48 VERBOSE[4193] logger.c: ChanSel: Reserved Jul 18 12:51:48 VERBOSE[4193] logger.c:Ext: 1 Coding: 0 Number Specified Channel Type: 3 Jul 18 12:51:48 VERBOSE[4193] logger.c
Re: [asterisk-users] IAX call limit
Gordon Henderson wrote: On Sun, 21 Jan 2007, Cristian Draghici wrote: IDEfisk is based on iaxclient (iaxclient.sourceforge.net) which will reject a call with the BUSY signal if there is no available line in the softphone to take the call. This means you need to configure IDEfisk to use only one line (call context). I don't know if this is possible. Somewhere in IDEfisk is this call that initialised iax client: iaxclient.h:EXPORT int iaxc_initialize(int audType, int nCalls); what you want is nCalls to be 1. There is a configurable parameter in sip.conf: call-limit, but this seems to be missing from the iax channel setup. Maybe this is deliberate for other reasons though. But switching to a SIP client and using this might work, if SIP is an option for you. Gordon Hope this helps, Cristi -- Cristian Draghici http://www.loudhush.ro On 1/21/07, Nir Simionovich [EMAIL PROTECTED] wrote: Hi Philipp, Thanks for the tip, but that is not what I initially meant. I'm using IDEfisk, and I would like it when a call comes Into IDEfisk to generate a BUSY signal, if there is already a call in the client. Any ideas ? Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Thursday, January 18, 2007 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX call limit Nir Simionovich wrote: Stupid and silly question - is there a way to limit the number of concurrent calls an IAX client can make? something in the similar sense of incominglimit and outgoing limit on SIP? It can be done in the dial plan: http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You may consider using the macro superdial. There you can specify the maximum number of concurrent calls per group, so that the next one recieves the busy tone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help - SOLVED -
Giorgio Incantalupo wrote: Hi Tommaso, have you tried to search for noise suppression? I remember some phone has a function to automatically suppress it so the caller does not hear anything and thinks the other party has hung up. Giorgio Incantalupo Tommaso Calosi wrote: I have this problem with Asterisk 1.2.4 I hear other party's voice only when I speack or i make some noise. Otherwise i hear nothing. Futhermore every time i receive a call , this message is displayed : -- Started music on hold, class 'my_class', on SIP/ some random public ip address -08222740 any help? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've solved right now. The problem was occurring just with musiconhold. Icoming calls are answerd and a message is played back using dial with m option. The problem is that the caller has silence unless it produces noise. I had to modprobe ztdummy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help
I have this problem with Asterisk 1.2.4 I hear other party's voice only when I speack or i make some noise. Otherwise i hear nothing. Futhermore every time i receive a call , this message is displayed : -- Started music on hold, class 'my_class', on SIP/ some random public ip address -08222740 any help? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call transfer asterisk + with SPA-1001
Does anybody knows how to transfer calls from Sipura SPA 1001 configured as asterisk internal ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with Junghanns Quadbri
I think you'll need to set the jumpers on the card in order to specify the NT ports. Jean-Louis curty wrote: Hi everybody I hope that somebody can help me with the following I have 2 quadbri cards 2 - 1t0 cards 1 pabx alcatel 4200 I would like to connect my asterisk to the alcatel , I installed bristuff 0.3.0-1p , loaded the zaphfc driver in NT mode configured zaptel and zapata , it works great. then I removed the 1 t0 card, added the quadbri loaded qozap : insmod qozap.ko ports=15 ( 4 ports in NT ) adjusted the zaptel zapata, specified the right signalling, right context ran ztcfg -vv ( 12 channels configured ) started asterisk, I get layer1 down message on the 4 ports, leds remain red what ever I do in my conf , I am not able to get a reaction from the card ( I tried with my two quadbri, on 2 different pc's ) what can I check ? thanks jl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Digium Card b410p
Who knows something interesting about the new BRI digium card b410p ? For example, will it use the misdn driver or the native zaptel? Any interesting links will be appreciated too. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Passsword Issue
I have had the same problem too, I solved resetting the phone to factory defaults Edward de Zeeuw wrote: I'll take a look first thing tomorrow and let you know what I find. Thanks! Edward Colin Anderson wrote: In the Snom web management page under Advanced make sure Challenge response on phone is turned to OFF. This is a stupid feature to have on by default from the factory. -Original Message- From: Edward de Zeeuw [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 21, 2006 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Snom 360 Passsword Issue I have multiple (20+) Snom 360 phones communicating with asterisk 1.2.7.1. Almost regularly (daily) and in some cases ongoing 9every 10 minutes the phones ask for password and id the account they are seeking the password for. If I hit the X key the phone continues operating normally. Has anyone else come across a similar issue? Edward ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call limit function on sip channel to external pop
you can either use the call_limit for each internal or if you wish something centralized ( for example a maximum total of 30 concurrent calls ) you can use the superdial macro http://www.voip-info.org/wiki/view/Superdial+macro Patrick wrote: On Tue, 2006-06-20 at 09:20 +0200, bram kortleven wrote: Hi, We've been using asterisk as our main telephone-communications platform for years now, and we wrote several extra scripts and features for it. Now we 're looking for a solution to limit the number of channels going to an external SIP provider. We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be able to use such features, but nothing helped... When we configure a new channel, it seems to work, but putting the call_limit on an existing sip channel going out, it doesn't do anything. Anyone already had such an issue, or anyone knowing the best config for limiting outgoing sip channels to external sip providers? Previous answers to similar questions usually point to: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk
I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the disable call waiting in asterisk as well, but I have not been able to find any documentation for this. I have found this http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting about call waiting, but it's quite unusefull. Thanks Tommaso Calosi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk
Well, it does help, but it causes the announced transfer to fail, because if you set call-limit=1 you cannot dial out to announce the transfer... Alberto Sagredo wrote: It has a conceptual problem i have notified several times to Cisco-Linksys. It could not be disabled, i have the same problem with my queue extensions, and the way to resolve has been to use call-limit=1 in extensions. i hope this helps. Tommaso Calosi escribió: I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the disable call waiting in asterisk as well, but I have not been able to find any documentation for this. I have found this http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting about call waiting, but it's quite unusefull. Thanks Tommaso Calosi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playback welcome message while phones ring, please help
I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback welcome message while phones ring, please help
Thanks, it works for me too. Tim Sharp wrote: Yes it does, I just set our system up that way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gareth Blades Sent: Tuesday, June 06, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback welcome message while phones ring,please help I believe if you use the new native music on hold feature it always plays the music on hold starting from the beginning. On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote: I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Guido Hecken wrote: I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. We had the same problems with some cheap LevelOne Switches. The Snoms rebooted during a call, calls dropped etc. Replacing the switches was the solution. Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have moved to 3com switches,but the Snom 320 still locks up, and also I don't think it's reasonable to force customers to buy 3com just because Snom firmware sucks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] misdn problem
In your extension.conf, in the misdn context you defined in /etc/asterisk/misdn.conf you have to add something linke the following line [from-pstn] exten = 0108680550,1,Dial(SIP/201) If you don't want to have to write a string for each called extension, you can put something like. Obviously SIP/201 is the phone you wish to ring. [from-pstn] exten = _.,1,Dial(SIP/201) [EMAIL PROTECTED] wrote: I have two HFC ISDN Cards, configured using mISDN on asterisk svn head 1.2 These two cards are connected to 2 ISDN Lines, receiving calls for 50 numbers. Everything is OK on 75 % and bad on 25 % When is bad, In /var/log/asterisk/full I see May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension can never match, so disconnecting May 26 09:55:28 DEBUG[24410] channel.c: Prodding channel 'mISDN/1-1' May 26 09:55:28 DEBUG[24410] channel.c: Scheduling timer at 160 sample intervals May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension can never match, so disconnecting May 26 09:55:28 DEBUG[24410] channel.c: Prodding channel 'mISDN/2-1' May 26 09:55:28 DEBUG[24410] channel.c: Scheduling timer at 160 sample intervals on the asterisk console, if I set misdn set debug 10, I see P[ 1] handle_frm: frm-addr:42000103 frm-prim:3f082 P[ 1] -- lib: NEW_CR Ind with l3id:200ec on this port. P[ 1] -- new_process: New L3Id: 200ec P[ 1] handle_frm: frm-addr:42000103 frm-prim:30582 P[ 1] set_channel: bc-channel:0 channel:1 P[ 1] lib Got Prim: Addr 42000103 prim 30582 dinfo 200ec P[ 1] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 0] $$$ find_chan: No channel found with l3id:200ec P[ 1] I IND :SETUP oad:3481303064 dad:0108680550 P[ 1] -- mode:TE cause:16 ocause:16 rad: P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE P[ 1] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 1] -- screen:0 -- pres:0 P[ 1] -- channel:1 caps:Speech pi:0 keypad: P[ 1] -- urate:0 rate:16 mode:0 user1:0 P[ 1] -- pid:336 addr:50010102 l3id:200ec P[ 1] -- b_stid:0 layer_id:50010180 P[ 1] -- bc:81681d4 h:0 sh:0 P[ 1] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 1] -- Bearer: Speech P[ 1] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:0108680550 oad:3481303064 P[ 1] read_config: Getting Config P[ 1] config_jb: Called P[ 1] -- * CallGrp: PickupGrp: P[ 1] * Queuing chan 0x842e8b0 P[ 1] CONTEXT:from-pstn P[ 1] Tone Indicate: P[ 1] -- Busy P[ 1] misdn_write: * prods us P[ 1] SENDEVENT: stack-nt:0 stack-uperid:4104 P[ 1] I SEND:DISCONNECT oad:3481303064 dad:00108680550 P[ 1] -- mode:TE cause:16 ocause:1 rad: P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE P[ 1] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 1] -- screen:0 -- pres:0 P[ 1] -- channel:1 caps:Speech pi:0 keypad: P[ 1] -- urate:0 rate:16 mode:0 user1:0 P[ 1] -- pid:336 addr:50010102 l3id:200ec P[ 1] -- b_stid:0 layer_id:50010180 P[ 1] -- bc:81681d4 h:0 sh:0 P[ 1] GOT SETUP OK P[ 1] Freeing Msg on prim:30582 P[ 2] handle_frm: frm-addr:42000203 frm-prim:3f082 P[ 2] -- lib: NEW_CR Ind with l3id:40076 on this port. P[ 2] -- new_process: New L3Id: 40076 P[ 2] handle_frm: frm-addr:42000203 frm-prim:30582 P[ 2] set_channel: bc-channel:0 channel:1 P[ 2] lib Got Prim: Addr 42000203 prim 30582 dinfo 40076 P[ 2] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 1] $$$ find_chan: No channel found with l3id:40076 P[ 2] I IND :SETUP oad:3481303064 dad:0108680550 P[ 2] -- mode:TE cause:16 ocause:16 rad: P[ 2] -- facility:FAC_NONE out_facility:FAC_NONE P[ 2] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 2] -- screen:0 -- pres:0 P[ 2] -- channel:1 caps:Speech pi:0 keypad: P[ 2] -- urate:0 rate:16 mode:0 user1:0 P[ 2] -- pid:337 addr:50010202 l3id:40076 P[ 2] -- b_stid:0 layer_id:50010280 P[ 2] -- bc:8173dec h:0 sh:0 P[ 2] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 2] -- Bearer: Speech P[ 2] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:0108680550 oad:3481303064 P[ 2] read_config: Getting Config P[ 2] config_jb: Called P[ 2] -- * CallGrp: PickupGrp: P[ 2] * Queuing chan 0x84a8ea0 P[ 2] CONTEXT:from-pstn P[ 2] Tone Indicate: P[ 2] -- Busy P[ 2] misdn_write: * prods us P[ 2] SENDEVENT: stack-nt:0 stack-uperid:4204 P[ 2] I SEND:DISCONNECT oad:3481303064 dad:00108680550 P[ 2] -- mode:TE cause:16 ocause:1 rad: P[ 2] -- facility:FAC_NONE out_facility:FAC_NONE P[ 2] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 2] -- screen:0 -- pres:0 P[ 2] -- channel:1 caps:Speech pi:0 keypad: P[ 2] -- urate:0 rate:16 mode:0 user1:0 P[ 2] -- pid:337 addr:50010202 l3id:40076 P[ 2] -- b_stid:0 layer_id:50010280 P[ 2] -- bc:8173dec h:0 sh:0 P[ 2] GOT SETUP OK P[ 2] Freeing Msg on prim:30582 P[ 2] MGMT: Short status dinfo 201 P[ 2] MGMT: SSTATUS: L2_ESTABLISH all the (4) channels were idle at call time. Thanks in advance Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it
Re: [Asterisk-Users] Snom firmwares suck
I agree... Snom firmware are buggy. You can use any version, you'll always find a bug, expecially for 320's. I think instead of developing the XML minibrowser they should make the phones to work properly. Dovid Bender wrote: I was transporting it in my suitcase when I flew from NY to FL. When I got there the screen was dead. I contacted snom and they were real helpfull and resolving the issue. Dovid */Alex Robar [EMAIL PROTECTED]/* wrote: Interesting, I've never had a screen just die on me. Are you saying that the screen just stops working? As for firmware, I've always found that the best way to deal with problematic firmware is to talk to the company about it. Especially in a scenario like you're in, where everyone else seems happy with the firmware. It's very likely that Snom may not be aware of the problems you are experiencing. I can't say that I've ever had a Snom reboot in the middle of a conversation. I don't think my phones have ever just rebooted when I haven't asked them to. I would certainly talk to Snom about this. Alex On 5/19/06, *Remco Barende* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries http://us.rd.yahoo.com/mail_us/taglines/postman9/*http://us.rd.yahoo.com/evt=39666/*http://messenger.yahoo.com/ for just 2¢/min with Yahoo! Messenger with Voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Misdn 0.2.1 BUSY tone
I have this problem on misdn 0.2.1: in extension.conf i have such a situation; [misdn_incoming] exten = 06786541,1,Dial(SIP/203) where SIP/203 is a GXP-2000. I want to make the 203 to answer just one call at the same time, so i've disabled the call waiting feature on the phone, but when I do this the caller does not hear the Busy tone, it receives the telco Network error tone. I want the caller to receive the busy tone when the called is busy . How can I do this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed
I've tested this problem pretty much in these days. - A ( Snom 320 ) put B caller on Hold - C calls A - A press the transfer button twice - Now B and C are connected ( makes no sense to me ) So, for one side there is a human error ( the operator should not press the transfer button twice ) BUT SNOM MAKES ALL MORE COMPLICATED!!!. I think this feature is just causing problem to most of the snom users. It would be a very good thing if snom could disable this feature or at least make it possible to disable from users. This behavior is 100% indipendent from the Call join on Xfer setting. This is the answer from snom Dear Tommaso Calosi, Is the initial problem still present with V5.5? However Call join on Xfer works like this: When this feature is turned to on, you will connect an incoming call to, for example, a colleague you already have on hold by pressing TRANSFER. You will not be able to pick one of severals calls on hold to transfer the call to. If this is your usual scenario, set this feature to off and use the function keys -/- to select the calls to be joined! Press TRANSFER and ENTER to join them. If you have only two calls involved you can use Call join on Xfer. Only if you except many simultaneous calls on HOLD do not use it. However none of the transfer scenarios is supposed to switch to conference mode. Any new observations you may report me again. Tommaso Calosi - SYSMIC SRL [EMAIL PROTECTED] wrote: Franklin Webb wrote: We use a large number of Snom 320's and we have this same problem even with Call join on Xfer set to off. I had not previously linked it decisively to the Snoms, but it sounds like that is likely our issue. We've had to stay at the 4.5 firmware because otherwise we get additional incomming calls when our reps put someone on hold to make a transfer. If I find any solution I'll be sure to share it with the list. Thanks for sharing your experiences, Frank Webb Assistant Project Leader Inter Media Marketing Solutions - Original Message - *From:* Alexander Lopez mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, May 04, 2006 7:35 AM *Subject:* RE: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed Under Advanced make sure this is set: Call join on Xfer (2 calls): to off *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Tommaso Calosi *Sent:* Thursday, May 04, 2006 4:02 AM *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Unwanted conference with snom320 and asterisk 1.07bristuffed I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a similar problem was fixed with the Snom320 Release 5.2 (http://www.snom.com/snom320_release_notes.html ) It says: fixed unwanted conference bug in offhook/enter during ringback with an incoming call BUT my phones are already running 5.2 firmware. Any idea? Am I the only one with this problem? Do you think is the usual buggy-snom firmware problem? Or it might be an Asterisk problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unwanted conference with snom320 and asterisk 1.07 bristuffed
Title: Messaggio Ihave 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a similar problem was fixed with the Snom320 Release 5.2 (http://www.snom.com/snom320_release_notes.html ) It says: fixed unwanted conference bug in offhook/enter during ringback with an incoming call BUT my phones are already running 5.2 firmware. Any idea? Am I the only one with this problem? Do you think is the usual buggy-snom firmware problem? Or it might be an Asterisk problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Snom 320 HOLD and TRANSFER not detected
Title: Messaggio
Hi
Franklin,
I've
downgraded the firmware to 4.5 but that didn't solve. The problem was in
sip.conf in the field "fromuser" which i set to Name Surname. If I set the
fromuser field that way it doesn't transfer or hold. If iI set it to N.Surneme (
without space ) it works. The strange thing is that on gxp2000 it works in both
ways.
Bye
and thanks
Tommaso Calosi
-Messaggio
originale-Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Franklin
WebbInviato: giovedì 27 aprile 2006 21.23A: Asterisk Users
Mailing List - Non-Commercial DiscussionOggetto: Re: [Asterisk-Users]
Snom 320 HOLD and TRANSFER not detected
Hi there Tommaso,
You've probably already tried this, but a reset
and reboot often fixes certain flakey behaviours on our Snom 320s. We've
had to revert back to the 4.5 firmware because of some issues with the 5.0 and
later, mostly that additional calls come in when our reps put customers on
hold.
I have one phone here still at 5.3 firmware and I
tested the hold key and that did not happen.
Assuming everything is identicle in the config it
could be bad phones, but we've seen a very small percentage of those.
Overall we've found the Snom320 to be an excellent phone for our
purposes.
Best of luck,
Franklin Webb
Assistant Project Leader
Inter-Media Marketing Solutions
- Original Message -
From:
Tommaso Calosi
To: asterisk-users@lists.digium.com
Sent: Thursday, April 27, 2006 2:03
PM
Subject: [Asterisk-Users] Snom 320 HOLD
and TRANSFER not detected
I have a
preoblem with my snom 320 phones. I have 5 snom phones installed and all of
them have 5.2 firmware. All have same settings in the advanced panel.
On2 phones when I press the hold or transfer key nothing happens and *
does not start the musiconhold. In the The hold and transfer keys are set as
F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work
ok.
Any
ideas?
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[Asterisk-Users] Snom 320 HOLD and TRANSFER not detected
Title: Messaggio I have a preoblem with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2 firmware. All have same settings in the advanced panel. On2 phones when I press the hold or transfer key nothing happens and * does not start the musiconhold. In the The hold and transfer keys are set as F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work ok. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one-waysilence during calls
Title: Messaggio My sip phones are connected to asterisk PBX 1.2.4. The PBX is connected to the provider through IAX2 connection. Sometimes randomly the voice is stopped and both caller and called don't hear the other's voice. During this silence period Asterisk is not logging any errors. This happen on incoming calls too, and incoming calls are through ISDN BRI lines Any idea? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on hold not working between SIP clients
Hi, I have problems in configuring music on hold between SIP clients: if I dial from an iax to a sip and then the iax place the call on hold, then the music starts. If the sip client places the call on hold, the music does not start and nothing is reported by the Asterisk console. I think it may depend on the sip.conf configuration, but I couldnt find docs about this. Currently Im using madplay for music, and I have configured the musiconhold.conf with this setting [classes] default = custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay --mono -R 8000 -A -20 --output=raw:- Thanks Tommaso Calosi ___ 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
