Re: [asterisk-users] Zaptel 1.2.26 problems
Hey Ira, I'm glad someone else noticed this. I found out this with 3-4 installs the hard way. Latest (or should I say last X releases) zaptel 1.2 have some strange bug on wctdm and will not working OK for incoming calls. The problem described is exactly what I experienced, if you unload and load zaptel modules sometimes incoming rings will get detected and asterisk picks up, but I would say about 5% of test calls get pickd up with the newer versions... I am able to reproduce this almost immediately after installeing latest 1.2 from branch (ie. SVN-branch-1.2-r120109M). I thought it had to do with one install (low ring voltage, etc) but its definitely a zaptel version issue. I went back to zaptel (ie. SVN-branch-1.2-r46964M) and the wctdm goes back to work OK for incoming calls. If anyone from digium sees this, I'd be glad to help any way I can so that you can debug the problem. BTW loading zaptel with debug=1 will not give any relevant info on incloming calls that wont 'ring' via dmesg. Regards, Lex On Sun, Jul 13, 2008 at 2:30 AM, Ira [EMAIL PROTECTED] wrote: Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls on channel 1 but incoming calls showed nothing on the console. Reverting to 1.21.1 set it all back to working. Zap show channel(s) showed the channels there and seemingly alive but no calls. Any suggestions on where to look? Ira And apologies if the latest isn't 1.2.26. I think that's what it was. I also upgraded to Asterisk 1.2.29 but I'm still using that with the older Zaptel and all is well. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted
Congrats on going forward with the project Moises. MFC/R2 support on chan_zap sounds great, looking forward on trying it out. Regards, Lex On Thu, Apr 24, 2008 at 3:03 PM, Moises Silva [EMAIL PROTECTED] wrote: Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4, and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4 than the original Steve driver in 1.2, and with better sound under heavy loads. The Asutunicall page can be found here: http://www.moythreads.com/astunicall/ Hum, wonder who this moy is hey wait, that's me! . Even when is in my plans to keep giving general maintenance to chan_unicall, my long term plan is to leave R2 support into chan_zap, so I would recommend to all users to try chan_zap R2 support, the more users we get the faster the driver will be stable enough to replace chan_unicall, the less headaches you will have (I hope). - Moy or Moisés Silva, same shit :-) -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Ruben, Contact support at digium they have a release on a firmware that fixes this and other issues with the VPMADT032. Apparently it comes on newer zaptel drivers. Good luck with your install. Lex On Mon, Apr 7, 2008 at 3:11 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ruben Zamora wrote: Hi, I have a same problem, last week i was working with TE120 with a little echo in some call, I replace the card with a TE122B ( Included Echo Cancelation VPMADT032) and there was no more echo in my call. But know i have de same probelm with my incoming audio stream gets clipped / dropped when you speak. Please contact Digium technical support about this. This is definitely something that we need to work with the vendor of the echo canceller IP about. Matthew Fredrickson Thanks Ruben Lex Lethol escribió: Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hanging up a 3-way conference when middle user hangs up
Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once the user is already on the phone with someone he called. 1. Press 'Flash' on the phone. Party B will now be placed on hold and you will hear a dial tone. 2. Dial party C's number or a pre-configured speed dial followed by '#', (you can engage in conversation). 3. Press 'Flash' to join both C and B to a single conference. 4. When you place the phone's handset on-hook, party B and party C will remain in conversation. After step 4, B and C remain on the conversation and I am looking for a way to disable this without disabling 3 way calling. Basically I am looking for a way to force asterisk hang up both B and C once the 'middle' user hangs up so this will not leave channels stuck on trunks without disconnect supervision. Anyone know how to accomplish this? Any comments appreciated. ___ --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] Hanging up a 3-way conference when middle user hangs up
As far as I know when I setup a 3-way on something like a cisco will disconnect everyone when the middle (person who setup the conference) hangs up. The problem I describe happens on ATAs and the like that uses flash to put on hold while setting up the second call. I am not sure about other phones other than cisco, polycom and a few others. Thanks! Ed On 1/7/07, Paul Hales [EMAIL PROTECTED] wrote: How does this compare to using the conference features on a SIP phone, say a Snom? I have used a Snom many times for an ad-hoc conference, without any troubles... PaulH On Sun, 2007-01-07 at 18:12 -0600, Lex Lethol wrote: Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once the user is already on the phone with someone he called. 1. Press 'Flash' on the phone. Party B will now be placed on hold and you will hear a dial tone. 2. Dial party C's number or a pre-configured speed dial followed by '#', (you can engage in conversation). 3. Press 'Flash' to join both C and B to a single conference. 4. When you place the phone's handset on-hook, party B and party C will remain in conversation. After step 4, B and C remain on the conversation and I am looking for a way to disable this without disabling 3 way calling. Basically I am looking for a way to force asterisk hang up both B and C once the 'middle' user hangs up so this will not leave channels stuck on trunks without disconnect supervision. Anyone know how to accomplish this? Any comments appreciated. ___ --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] Asterisk + Orion E1 GSM Gateway
Hi yusuf, I am working right now on a similar setup. If its the PRI type theres not so much on the syncing part. You need the PRI crossover rj45, theres info on voip-info on that and Orion has software to configure via Serial cable the E1 PRI as NET/USER and Time syncs. I setup mine via zaptel using css,hdb3,crc on the span. I am still debugging outogoing traffic but incoming is working OK. Lex On 12/18/06, yusuf [EMAIL PROTECTED] wrote: Leo Ann Boon wrote: yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. Leo Hi, crazy thing is I dont have any manual or anything, just the Gateway. From reading the 'sales' doc on the Orion site, this is a PRI/Q.SIg type. But I dont have anything else besides that. I dont even know how to get the Serial cable to work to configure the Gateway (through Minicom/Hyperterminal, there is a configuration on Orion, or so I'm told.) Can you help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] PRI debugging outgoing not working, help needed
Hi, Ive been playing on a asterisk to orion gsm box E1 pri setup. I have achieved incoming calls to be passed to my asterisk box successfully but outgoing calls will just I have tried playing with various pridialplan and overlapdial settings and with no success. If anyone can make more sense from the log, I'd certainly appreciate it. I am sending a 10 digit number to be dialed. I guessed that since my mobile will take the number and dial without a problem it became my starting setup. zapata settings follow: switchtype=euroisdn pridialplan=unknown (tried with local) prilocaldialplan=unknown (tried with local) overlapdial=no (tried with yes) signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callerid=asreceived group=2 channel=32-46,48-62 Here is asterisk log with pri debuggin on: -- Executing Dial(SIP/301-9a6f, ZAP/32/xx|120|) in new stack -- Making new call for cr 32774 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0e 00 81 35 32 38 31 31 32 31 32 34 36 32 34] Calling Number (len=16) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number passed network screening (1) 'xx' ] [70 0d 80 35 32 38 31 31 32 31 32 34 36 32 34] Called Number (len=15) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'xx' ] [a1]xes*CLI Sending Complete (len= 1) -- Called 32/xx Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) -- Zap/32-1 is proceeding passing it to SIP/301-9a6f Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CONNECT (7) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 84 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/32-1 answered SIP/301-9a6f Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: DISCONNECT (69) [08 02 84 f5] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Unknown (117), class = Interworking (7) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup request NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: RELEASE (77) [08 02 81 f5] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (117), class = Interworking (7) ] -- Hungup 'Zap/32-1' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/301-9a6f' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/301-9a6f' Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 84 f5] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Unknown (117), class = Interworking (7) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG:
[Asterisk-Users] Cisco 7960 mic generating noise on other end
Hi, I'm having a problem with one of our 7960. They all run latest 7.4 SIP firmware. The problem appears on the other end. The other end constantly hears a 'crackling' noise. I have tested using phone set, headset and speaker and the noise appears on all cases. I have other 7960 setup exactly same way (using same asterisk, firmware, etc) so it looks like a hardware issue. I'd appreciate if anyone has any insight on this or any other similar issues before I open the thing. Thanks! Lex ___ 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] new cisco ip video phone?
Hi, Just finished watching the season finale of '24' the TV series. Throughout the series they have been showcasing Cisco hardware especially Cisco IP phones (7970's). On the last episode or two they showed what seemed to me a new cisco IP video phone. It stands just as a 12 lcd screen with the cisco branding/logo and letters just as the 79xx series. I wonder if this is a new cisco model thats ready to roll out. It looks great, but then again, I doubt they will support SIP on it (at least on release) Anyone else know anything on this? Lethol ___ 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] new cisco ip video phone?
Hey all, I took some screenshots of the video feed... If you look closely at the upper right part of the 'phone' it says Cisco IP Phone and barely a 7xxx something. It may be a fake just for showing cisco capable of doing IP telephony but who knows.. decide for yourself ;) http://lethol.com/blog/ciscoIP.jpg http://lethol.com/blog/ciscoIP2.jpg http://lethol.com/blog/ciscoIP3.jpg http://lethol.com/blog/ciscoIP4.jpg http://lethol.com/blog/ciscoIP5.jpg http://lethol.com/blog/ciscoIP6.jpg Lethol On 5/26/05, Mailing List [EMAIL PROTECTED] wrote: Any chance it's the phone mentioned here? http://voxilla.com/voxstory134.html _ Mobilcom http://www.mobilcom.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol Sent: Thursday, May 26, 2005 2:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] new cisco ip video phone? Hi, Just finished watching the season finale of '24' the TV series. Throughout the series they have been showcasing Cisco hardware especially Cisco IP phones (7970's). On the last episode or two they showed what seemed to me a new cisco IP video phone. It stands just as a 12 lcd screen with the cisco branding/logo and letters just as the 79xx series. I wonder if this is a new cisco model thats ready to roll out. It looks great, but then again, I doubt they will support SIP on it (at least on release) Anyone else know anything on this? Lethol ___ 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.cc / Sixtel?
Ive been using it too and its working great. Still waiting for my DID but as far as terminating to the US I am very impressed with sound quality. Lethol On Mon, 13 Dec 2004 16:51:47 -0800 (PST), Steve Edwards [EMAIL PROTECTED] wrote: I've used them for a couple of months. My usage is very small, but I'm really impressed. Especially compared to VoicePulse. With Sixtel, when you call tech support, you get to talk to a person. That person actually knows what they are doing. With VoicePulse, I could never talk to a person and email would take days to get a response. Sixtel has great rates -- US$1.49 a month and US$.0137 a minute for a DID. VoicePulse was about US$8.00 a month and US$.03 a minute. Sixtel feels small and mom and popish. Getting a DID takes a couple of days and their web site is less complete and polished. VoicePulse DID's are active immediately I point anybody who asks to Sixtel. On Mon, 13 Dec 2004, Me wrote: Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO setup
So, if zaptel will not read codes from my indications conf file, what would be a suitable solution to feed it my country tones? Is there a list of loadzone/defaultzone country codes included in zonedata.c? Can this list be updated to include my country? (I have the tones) Thanks for the help Lethol On Wed, 17 Nov 2004 10:37:26 +0800, Dinesh Nair [EMAIL PROTECTED] wrote: On 17/11/2004 04:33 Matt Riddell said the following: Dinesh Nair wrote: doesn't it pull it from the structures hardcoded into zonedata.c ? iianm, indications.conf is only used for PlayTones(). Don't know, but I have non-standard tones defined which I analysed in Wavelab, and then added to indications.conf under my own entry. Before doing this, my line would never detect hangup. After I did it, there was no problem. i just checked the zaptel code, ztcfg.c specifically loads the loadzone/defaultzone tones from zonedata.c, it doesn't read indications.conf at all. i doubt the tonezones referred to in zaptel.conf are tied to the indications.conf definitions. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO setup
Does anyone know if this needs any special modification to work outside the US? I have setup my country's correct tone info and tested thru the indication.conf file. Question would be, where does my zaptel device get the tones expected for the busydetect procedure? How can I modify them? Is this the same as the callprogress action? Thanks to anyone who can guide/point me with the right direction on this! Lethol On Mon, 15 Nov 2004 15:56:44 -0500, Darly Coupet [EMAIL PROTECTED] wrote: Hi, Works as advised! Thanks Darly On 16 Nov 2004 at 8:26, Matt Riddell wrote: Darly Coupet wrote: Hi, Thanks for your response. More info as requested: Location: USA FXO connection: Wipphone.com service (similar to Vonage) Analog Telephone Adaptor: Webphone WP200 FXO Card: X100P * */etc/zaptel.conf /fxsks=1 # X100P defaultzone=us loadzone=us / /etc/asterisk/zapata.conf /signalling=fxs_ks ; X100P echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=default ; Points to the default context of your extensions.conf channel = 1 / You are missing: busydetect=yes busycount=10 from your zapata.conf file. Just make sure they are above the channel = 1 line. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Hang Problem
I am getting a system hang ups as well.. But my system will halt after being about 10 minutes on a call.. and there is no error showing up on asterisk CLI :S My linux server will just freeze and will only happen while on asterisk I have no ida on how to debug this one. I think it might ne a hardware compatibility problem with the OpenLine4 I am using. Any suggestions?? Lethol On Mon, 11 Oct 2004 19:38:41 +0200, Michael Bielicki [EMAIL PROTECTED] wrote: You have to adjust the file-max value for your kernel. If you use sysctl the setting is in the fs.file-max setting. cheers Michael On Mon, 11 Oct 2004 10:48:21 -0400, Darren Sessions [EMAIL PROTECTED] wrote: I am getting some weird behavior and a rash of interesting messages in the log files. If anyone has some ideas, it would be appreciated. Using Asterisk v1.0.1 on Suse Enterprise Linux v8.0. HP DL380 Server. 4GB Ram - Dual 3.2ghz processors. This first entry is when asterisk simply goes unresponsive. We've got a script that automatically polls asterisk (via sip) and restarts it if it does not receive a response. Notice the 9:56 to 10:01 gap. Oct 11 09:53:29 WARNING[6427661]: Failed to write frame Oct 11 09:55:53 WARNING[6445068]: Failed to write frame Oct 11 09:56:10 WARNING[6449163]: Failed to write frame Oct 11 10:01:59 NOTICE[6478861]: Removed default indication country 'us' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:02:01 NOTICE[1024]: parking.conf is deprecated in favor of 'features.c We've started getting allot of these messages in our log files. Unlikely that this is not associated with the first problem. Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:05 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:05 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys PAP2-NA
Im also interested in a couple of these... plesase email me if you are selling or post over a link! Lethol On Fri, 24 Sep 2004 22:02:52 -0400, William Suffill [EMAIL PROTECTED] wrote: Anyone here have any pointers of where to get 1 of the PAP2-NA. Given all the talk about it I'd be curious as to testing one myself . -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0 released
Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Softphone for PocketPC or iPaq
I tried the xten one and didn;t like at all.. Havent tried to SJPhone, but my guess is that it has better support. Lethol On Thu, 23 Sep 2004 08:13:10 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On 22 Sep 2004, Sudhir Kumar wrote: Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Xten has a product, possibly still in beta. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 released
Kenneth, Did you submit to slashdot and are you on Astricon?? Mark has just stated he will give out a price to the person who submitted to slashdot.. My submition got rejected :( You beat me to the minute. ;) Congrats if you did! Lethol On Thu, 23 Sep 2004 07:58:31 -0700, Kenneth Shaw [EMAIL PROTECTED] wrote: To be Slashdotted within 30 minutes. -Ken Shaw... On Thu, 2004-09-23 at 07:28, Lex Lethol wrote: Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP phone
Stay away from the 7910 if your going SIP. It will not support it. Lethol On Thu, 23 Sep 2004 01:45:23 +1000, Shaun Ewing [EMAIL PROTECTED] wrote: The 7910 does not support SIP. It is SCCP only. -Shaun - Original Message - From: Henry Devito [EMAIL PROTECTED] Date: Wed, 22 Sep 2004 10:44:02 -0500 Subject: [Asterisk-Users] Cisco IP phone To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Hi all, I have a person trying to sell me Cisco 7910 IP Phones. Does anyone know if SIP is supported on these phones? I have CCO login also so if they do support SIP does anyone know where I could download the software? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix OpenSwitch12
hi Flynn, I have an OpenLine4 on my setup. Everything appears to work finw and I am not having the hangup detect but I am having problems when voicemail tries to record via vpb channel. Did you ever have that on your OpenLine4? I have not tried out the OpenSwitch12 but I am a bit scared with voicetronix due to the lack of support and friendliness when debugging any problem that comes up. Just my 2 cents. Lethol On Mon, 06 Sep 2004 16:01:05 +0800, el Flynn [EMAIL PROTECTED] wrote: Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Telephony with Asterisk book
It definitely sounded sarcastic :P Lethol On Mon, 30 Aug 2004 08:21:06 -0400, Leif Madsen [EMAIL PROTECTED] wrote: On Mon, 30 Aug 2004 10:21:55 +0800, Joseph Shi [EMAIL PROTECTED] wrote: Steve Underwood Wrote: Just wait for the simplified Chinese version to appear in Shenzhen's Book City. :-) That's great! Will it have the English version as well? Any idea when it will be there? I think he was being sarcastic :) Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceronix and asterisk
Heya Kelvin, Are you using the latest asterisk download from voicetronix webpage. I got most asterisk features working with an OpenLine4 but I still have some bugs/incompatibility issues to resolve. Make sure you download the latest driver and asterisk and make. After installing the voicetronix driver make sure you do the ./echo test included on the README to be sure driver was correctly installed. Lethol On Tue, 31 Aug 2004 00:36:37 +1000, Kelvin And Lisa [EMAIL PROTECTED] wrote: I have installed a 6VPCI card from voicetronix's but i can't get astersik to use it! Now looking at the loaded modules the chan_vpb is not loaded- so I assume that is why it is not working. Now I modified my vpb.conf file and extensions.conf, have I missed something Has anyone a installation guide as I am very new to this!! I have had asterisk working with SIP extensions. by dowloading and making the following Zaptel Libpri asterisk. but after installing the driver for the voicetronix I get errors with the Zaptel when I make it #error modules should never use kernal system header files and the like?? Thanks kelvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix OpenLine4 immediately hangs up on every call
Benjk, I dont have an answer to your problem, but I am currently using the same asterisk CVS HEAD found in voicetronix webpage. Most features are working OK and I am currently trying fo fix a voicemail problem but it appears not to be related to loopdrop. Are you sure the card works fine? (hardware wise) I modded the useloopdrop flag but I have no way of really testing it to see any difference. Make sure you run same context on the vpb.conf and somewhere in your extension.config. II know it sounds newbie-ish, but I am a newbie on asterisk and actually have been finding out the hard way on how to get things working. Anyway, good luck getting it to work. If it does maybe you can try out voicemail from a vpb channel (thats the current problem I am facing) :P Lethol On Mon, 30 Aug 2004 18:33:49 +0900 (JST), Sunrise Ltd [EMAIL PROTECTED] wrote: Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to the same lines. This happens both with UseLoopDrop = 0 and 1 settings in vpb.conf. There don't seem to be any other parameters in the conf file to control this. Has anybody else experienced this? Does anybody know how to teach the vpb driver to behave? BTW, the card is supposed to work in Japan. The console log is provided below... vpb/1-4: chanreads: Got Asterisk bridge with [SIP/2062-70de]. vpb/1-4: chanreads: Checking dtmf's vpb/1-4: chanreads: getting buffer! vpb/1-4: chanreads: got buffer! vpb/1-4: chanreads: applied gain vpb/1-4: chanreads: queueing buffer on read frame q (state[6]) vpb/1-4: Read channel (codec=0) -12 3 vpb/1-4: chanreads: Finished cycle... vpb/1-4: chanreads: Starting cycle ... vpb/1-4: chanreads: Checking bridge vpb/1-4: chanreads: No native bridge. vpb/1-4: chanreads: Got Asterisk bridge with [SIP/2062-70de]. vpb/1-4: chanreads: Checking dtmf's vpb/1-4: chanreads: getting buffer! vpb/1-4: Event [12=[03] Loop Drop] vpb/1-4: Flushing event [12]=[03] Loop Drop vpb/1-4: handle_owned: got event: [12=0] vpb/1-4: handle_owned: putting frame type[4]subclass[1], bridge=(nil) == vpb/1-4: Hangup requested vpb/1-4: chanreads: got buffer! vpb/1-4: chanreads: applied gain vpb/1-4: p-stopreads[1] p-owner[0x8109238] vpb/1-4: chanreads: Finished cycle... == vpb/1-4: Ending record mode (1/yes) vpb/1-4: stopped record thread on vpb/1-4 == vpb/1-4: Ending play mode on vpb/1-4 vpb/1-4: Setting state down == vpb/1-4: Hangup complete Restarting monitor Trying to reawake monitor Monitor restarted == Spawn extension (Internal, 809061554123, 2) exited non-zero on 'SIP/2062-70de' Monitor got null event vpb/1-4: Event [12=[03] Loop Drop] vpb/1-4: Flushing event [12]=[03] Loop Drop vpb/1-4: handle_notowned: mode=3, event[12][[03] Loop Drop ]=[0] vpb/1-4: handle_notowned: mode=3, [12=0] thanks in advance regards benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help debugging voicemail problem
Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that channel and outputs the following: -- Executing VoiceMail(vpb/1-1, u3001) in new stack -- Playing 'voicemail/default/3001/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/3001/INBOX/msg0001 format: wav49, 0x8149590 -- x=1, open writing: /var/spool/asterisk/voicemail/default/3001/INBOX/msg0001 format: gsm, 0x81496b0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/3001/INBOX/msg0001 format: wav, 0x81497c0 Aug 30 00:05:07 WARNING[19475]: app_voicemail.c:1442 play_and_record: No audio available on vpb/1-1?? -- User hung up -- Executing Hangup(vpb/1-1, ) in new stack == Spawn extension (incoming-pstn, 3001, 4) exited non-zero on 'vpb/1-1' == vpb/1-1: Hangup requested == vpb/1-1: Ending record mode (1/yes) vpb/1-1: stopped record thread on vpb/1-1 == vpb/1-1: Ending play mode on vpb/1-1 vpb/1-1: Setting state down == vpb/1-1: Hangup complete Restarting monitor Trying to reawake monitor Monitor restarted Monitor got null event Any advice/pointers/suggestion are greatly appreciated :) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicetronix Segmentation Fault
Hi, I am using a voicetronix OpenLine4. I downloaded a recent asterisk CVS from voicetronix webpage but have had no luck to reduce echo on outgoing calls and for it not to crach on incoming calls. I dont think both problems are related though. Here is an output of what happens when a new call comes in and my voicetronix tries to pick it up and crashes asterisk: vpb/1-1: Event [0=[00] Ring] vpb/1-1: handle_notowned: mode=3, event[0][[00] Ring ]=[0] vpb/1-1: New call for context [pstn] Aug 27 09:06:11 WARNING[19475]: pbx.c:1868 ast_pbx_run: Channel 'vpb/1-1' sent into invalid extension 's' in context 'default', but no invalid handler == vpb/1-1: Hangup requested vpb/1-1: Setting state down CID record - start vpb/1-1: Flushing event [11]=[00] Ring Off == vpb/1-1: Hangup complete Restarting monitor Trying to reawake monitor Monitor restarted CID record - skipped 602.460051ms trailing ring CID record - recorded 1711.737009ms between rings Segmentation fault Any advice on how to correct this or the other problem would be appreciated. ;) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 - SCCP or SIP?
On my experience, you should go to SIP whenever possible. 7940/60 on SIP will do most if not all fuctions. Try the little chart on support hardware on chan-sccp.sourceforge.net Lethol - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: Fri, 27 Aug 2004 14:16:11 +0100 Subject: [Asterisk-Users] Cisco 7940 - SCCP or SIP? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems this is where some sort of black magic needs to be used as I cannot find any way of getting them to work which leads me to the main question Is it better to use chan_sccp or SIP? I know these phones can work in either mode I was just wandering which is the better format and which has the most functions implemented? Its a simple home environment that I am planning but it would be good to be able to use the softkeys to transfer calls and to pickup messages. Thanks in advance, Sam Kevin Walsh [EMAIL PROTECTED] wrote on 27/08/2004 13:59:09: Michael Manousos [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Michael Manousos [EMAIL PROTECTED] wrote: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. Well, no matter what kind of concealment algorithm is used, just the first one or two packets will be concealed. The rest losses will result in no-playback. No CN interpretation, just absolute silence. That's true - unless there's some logic to say that after x lost packets, the line state should switch to CN generation instead of silence. The line state would switch back once a fresh RTP packet is received. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. Exactly. So the receiver, in case of no-receiption, should go back and see what was the last packet correctly received and act as I described above. Maintaining an audio state flag (CN/RTP) would be the key here. The difficult part to handle would be late or out-of-sequence RTP packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. Actually this is not so difficult, if there is a jitter buffer. Right. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Winckworth Sherwood Solicitors and Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Great Peter Street, London SW1P 3LR Telephone 020 7593 5000 Fax 020 7593 5099 Confidentiality This email message and any attachments are confidential; they may be subject to legal professional privilege and are intended for the named recipient only. If you are not the named recipient, please return the message and enclosures immediately and delete them from your system. Caution Before advice received only by email (whether by attachment or otherwise) may be relied on, the authenticity of the communication must be verified by means independent of email. Regulation The firm is regulated by the Law Society. Partners A list of partners is available for inspection at each office of the firm and on the firm's website at www.winckworths.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
Re: [Asterisk-Users] cisco phones w/ asterisk
Make sure not to buy any 7910 if you want an all SIP network. I dont see any advantages for it specially if you are in a planning stage. I also have seen tons of posts sayin that the 7920's are a pain to get 'em working. Lethol On Wed, 18 Aug 2004 17:06:17 -0700, Scott Laird [EMAIL PROTECTED] wrote: On Aug 18, 2004, at 4:54 PM, Chris Verges wrote: Chris Shaw wrote: The power of Christ compels thee Not to buy Cisco... hehe J/K don't do that. my employer wouldn't like me for poking fun at their products. :-P actually, i'm planning an asterisk-based voip network and was thinking of using the 7940/7920 phones for the end stations. i've heard both good and bad things from word-of-mouth, so now i pose the question here. good idea to do? My 7940 works flawlessly. It's running 6.3. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem compiling chan_sccp
Hi Julien, Thanks for the feedback. I am currently trying to compile with gcc-2.96-113 I've been trying all day to get CVS from sf.net to try to compile from latest version. Hope they fix it soon :S. Lethol On Tue, 17 Aug 2004 16:35:11 +1000, Julien Goodwin [EMAIL PROTECTED] wrote: On Mon, Aug 16, 2004 at 04:46:56PM -0500, Lex Lethol arranged a set of bits into the following: I recently bought a 7910. I found out too late that it would not do SIP as I initially thought. Anyway before ditchingit for a 7960 I wanted to try it out, I read that the guys at http://chan-sccp.sourceforge.net/ had done some improvements to the original chan_sccp driver and having 80% functionality with this model. I have not been able to compile their driver and keep getting the following: chan_sccp$ make Now compiling sccp_channel.c 264 lines sccp_channel.c: In function `sccp_channel_endcall': sccp_channel.c:234: parse error before `timer' sccp_channel.c:237: `r1' undeclared (first use in this function) sccp_channel.c:237: (Each undeclared identifier is reported only once sccp_channel.c:237: for each function it appears in.) sccp_channel.c:238: `cmtime' undeclared (first use in this function) make: *** [.tmp/sccp_channel.o] Error 1 Are you running GCC 2.95? If so there might still be a few cleanup patches to fix compilation that haven't hit CVS yet. (And unfortunatly anon CVS is down at the moment thanks to SourceForge...) Take a look at my patch available at: http://www.czmok.de/devtrack/bug_view_advanced_page.php?bug_id=033 (I run chan_sccp for cisco 12SP phones and that was the patch I needed to get it working under GCC 2.95) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem compiling chan_sccp
Julien, Just to let you know that I manually included your patch and everything compiled OK. I'll begin testing now. Thanks! Lethol On Tue, 17 Aug 2004 02:02:58 -0600, Lex Lethol [EMAIL PROTECTED] wrote: Hi Julien, Thanks for the feedback. I am currently trying to compile with gcc-2.96-113 I've been trying all day to get CVS from sf.net to try to compile from latest version. Hope they fix it soon :S. Lethol On Tue, 17 Aug 2004 16:35:11 +1000, Julien Goodwin [EMAIL PROTECTED] wrote: On Mon, Aug 16, 2004 at 04:46:56PM -0500, Lex Lethol arranged a set of bits into the following: I recently bought a 7910. I found out too late that it would not do SIP as I initially thought. Anyway before ditchingit for a 7960 I wanted to try it out, I read that the guys at http://chan-sccp.sourceforge.net/ had done some improvements to the original chan_sccp driver and having 80% functionality with this model. I have not been able to compile their driver and keep getting the following: chan_sccp$ make Now compiling sccp_channel.c 264 lines sccp_channel.c: In function `sccp_channel_endcall': sccp_channel.c:234: parse error before `timer' sccp_channel.c:237: `r1' undeclared (first use in this function) sccp_channel.c:237: (Each undeclared identifier is reported only once sccp_channel.c:237: for each function it appears in.) sccp_channel.c:238: `cmtime' undeclared (first use in this function) make: *** [.tmp/sccp_channel.o] Error 1 Are you running GCC 2.95? If so there might still be a few cleanup patches to fix compilation that haven't hit CVS yet. (And unfortunatly anon CVS is down at the moment thanks to SourceForge...) Take a look at my patch available at: http://www.czmok.de/devtrack/bug_view_advanced_page.php?bug_id=033 (I run chan_sccp for cisco 12SP phones and that was the patch I needed to get it working under GCC 2.95) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling chan_sccp
Hi, I recently bought a 7910. I found out too late that it would not do SIP as I initially thought. Anyway before ditchingit for a 7960 I wanted to try it out, I read that the guys at http://chan-sccp.sourceforge.net/ had done some improvements to the original chan_sccp driver and having 80% functionality with this model. I have not been able to compile their driver and keep getting the following: chan_sccp$ make Now compiling sccp_channel.c 264 lines sccp_channel.c: In function `sccp_channel_endcall': sccp_channel.c:234: parse error before `timer' sccp_channel.c:237: `r1' undeclared (first use in this function) sccp_channel.c:237: (Each undeclared identifier is reported only once sccp_channel.c:237: for each function it appears in.) sccp_channel.c:238: `cmtime' undeclared (first use in this function) make: *** [.tmp/sccp_channel.o] Error 1 The README only instructs to change the path to asterisk source which is /usr/src/asterisk in my case. I have a recent asterisk installed running OK as far as I know, but I have not been able to compile this. Any feedback would be greatly appreciated. Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling chan_sccp
Hi, I recently bought a 7910. I found out too late that it would not do SIP as I initially thought. Anyway before ditchingit for a 7960 I wanted to try it out, I read that the guys at http://chan-sccp.sourceforge.net/ had done some improvements to the original chan_sccp driver and having 80% functionality with this model. I have not been able to compile their driver and keep getting the following: chan_sccp$ make Now compiling sccp_channel.c 264 lines sccp_channel.c: In function `sccp_channel_endcall': sccp_channel.c:234: parse error before `timer' sccp_channel.c:237: `r1' undeclared (first use in this function) sccp_channel.c:237: (Each undeclared identifier is reported only once sccp_channel.c:237: for each function it appears in.) sccp_channel.c:238: `cmtime' undeclared (first use in this function) make: *** [.tmp/sccp_channel.o] Error 1 The README only instructs to change the path to asterisk source which is /usr/src/asterisk in my case. I have a recent asterisk installed running OK as far as I know, but I have not been able to compile this. Any feedback would be greatly appreciated. Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
