Re: [asterisk-users] detecting busy on queue transfer
That's my fault in the example - I forgot to add in the j. Anyway what
is strange is that I get my dialplan to jump to position 108, but at that
point the agent is disconnected. I thought that when falling out of the
queuetransfer context, the control would be returned to the trasferer,
after hearing the I'm sorry tone. Anything I'm missing here?
l.
In data Mon, 02 Oct 2006 00:36:30 +0200, Marco Mouta
[EMAIL PROTECTED] ha scritto:
Hi,
I've been looking the application dial on my asterisk server 1.2.9, and
as
far
CLI show application Dial
j- Jump to priority n+101 if all of the requested channels were busy.
It means that the application Dial on Asterisk 1.2 doesn't jump
automatically on Busy to the extension n+101, only if you Dial it with
j
argument!
exten = _0.,7,Dial(Zap/g1/${EXTEN:1},,j)
exten = _0.,108,NoOp(Got busy here)
Or you should handle it on you priority 8 in your dialplan
exten = _0.,7,Dial(Zap/g1/${EXTEN:1})
exten = _0.,8,Goto(s-${DIALSTATUS},1)
This is just an example.
Hope it helps, please give me some feeback.
On 10/1/06, Lenz [EMAIL PROTECTED] wrote:
I don't think that is the case - if I add a wait(10) after the step 108,
i.e. the busy detection, the agent seems to be disconnected immediately
at
the dial(), not after 10 seconds. That is what made me wonder what was
going on.
Yours
l.
On Sun, 01 Oct 2006 17:49:15 +0200, Adam Goryachev
[EMAIL PROTECTED] wrote:
Try adding this line:
exten = _0.,109,Hangup
Dunno if it will solve it, but might help :)
Regards,
Adam
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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[asterisk-users] Bristuff vs. vISDN vs. mISDN for hfc card ?
Hi, some time ago we used bristuffed Asterisk for our hfc cards cause it offered more features (echo cancellation most important) and was quite stable... I'm seeing now (I'm putting together Asterisk after a long time with hfc card) that there are now 3 choices for hfc chipsets : vISDN, mISDN and bristuff. What are pros and cons of each of them and what do you put in your Asterisks ? How they differ in functionality ? Thanks in advance, regards, Rob. ___ --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] channel_find_locked
Hi. Could you help me with this warning? channel_find_locked: Avoided initial deadlock for '0x8218ac0', 10 retries!] I have no ideea what causes it... It seems to appear only when i make a call... ___ --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] Bristuff vs. vISDN vs. mISDN for hfc card ?
On 08:55, Mon 02 Oct 06, Robert Rozman wrote: Hi, some time ago we used bristuffed Asterisk for our hfc cards cause it offered more features (echo cancellation most important) and was quite stable... I'm seeing now (I'm putting together Asterisk after a long time with hfc card) that there are now 3 choices for hfc chipsets : vISDN, mISDN and bristuff. What are pros and cons of each of them and what do you put in your Asterisks ? How they differ in functionality ? mISDN has the pro that chan_misdn is part of the default asterisk. I have no idea if the mISDN drivers are part of the default kernel, so you might check that. Bristuff has the pro that it comes in one package with a nice ./install script. This makes it easy to install everything you need. vISDN I dont know about. Maybe others can comment on that. I prefer to use bristuff, but that's because we have some installs with quodbri cards and we want to have the same version on all boxes. We dont use the hfc part, because of the quality compared to the more robust cards (quadbri, eicon etc) my 2 cents -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] cisco 2600
I've got the same question actually. We're looking to replace CCM with * (finally.. it took me ages to convince that * is way better), but we've got cisco 1700 2600 gateway's for the CCM in our remote offices that would have to be used by SIP with * now. Did anyone ever encounter or set up such an environment? Is it viable or should I go for a centralised setup in the head office straight away. greets Tijl Van den Broeck On 8/5/06, FaberK [EMAIL PROTECTED] wrote: Hi, does anybody used cisco 2600 as * gateway with E1? Thanks -- .:FaberK:. ___ --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] cisco 2600
We've been using cisco 2600 gateways with asterisk for a year and everything works fine. IOS 12.2 is installed in gateways. -Original Message- From: Tijl Van den Broeck [mailto:[EMAIL PROTECTED] Sent: Monday, October 02, 2006 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco 2600 I've got the same question actually. We're looking to replace CCM with * (finally.. it took me ages to convince that * is way better), but we've got cisco 1700 2600 gateway's for the CCM in our remote offices that would have to be used by SIP with * now. Did anyone ever encounter or set up such an environment? Is it viable or should I go for a centralised setup in the head office straight away. greets Tijl Van den Broeck On 8/5/06, FaberK [EMAIL PROTECTED] wrote: Hi, does anybody used cisco 2600 as * gateway with E1? Thanks -- .:FaberK:. ___ --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
I: [asterisk-users] Sip answer one side , ring other side
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di antonioInviato: sabato 30 settembre 2006 17.27A: asterisk-users@lists.digium.comOggetto: Re: [asterisk-users] Sip answer one side , ring other side when i make the call , on the xlite side i see the call connected but for the sip gateway the call is ringing and even the phone (PSTN side) is ringing. I thing that is only Asterisk send to xlite the signal of connect . Is there any configuration to set ?? Thanks Date: Sat, 30 Sep 2006 08:21:21 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Sip answer one side , ring other side To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed antonio wrote: Hi, the scheme is this : xlite --- Asterisk --- SIP gateway --- PSTN When i make a call with xlite (sip) to asterisk on the display of xlite i see that the call is connected but the phone is still ringing .. You must configure your gateway to NOT answer the call before making the PSTN call. Some gateway call that '1-step' dialing. ___ --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] Siemens Hipath - asterisk, pri problem
Hi all,i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones.asterisk (te110p) -- (TMS2) hipath 3550 in the future i'll connect the hipath to a telecom pri. the pri in the hipath is configured as EURO PP (with CRC4)i've already checked all previust post about this topic without any clue.many thanks/etc/zaptel.confspan=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15dchan=16bchan=17-31loadzone=itdefaultzone=it/etc/asterisk/zapata.conf[trunkgroups][channels]language=itcontext=from-zaptel signalling=pri_netswitchtype=euroisdnrxwink=300 ; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines;;usedistinctiveringdetection=yes usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=no echotraining=800rxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nochannel = 1-15channel = 17-31---thats the debug when i try to call from the hipath Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1 -- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification)-- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null --byebivio ___ --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] Siemens Hipath - asterisk, pri problem
Look at your extensionsincontext "from-zaptel" adding the s extensionsand add immediate=yes in zapata.conf Ciao Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di bivioInviato: lunedì 2 ottobre 2006 10.04A: asterisk-users@lists.digium.comOggetto: [asterisk-users] Siemens Hipath - asterisk, pri problem Hi all,i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones.asterisk (te110p) -- (TMS2) hipath 3550 in the future i'll connect the hipath to a telecom pri. the pri in the hipath is configured as EURO PP (with CRC4)i've already checked all previust post about this topic without any clue.many thanks/etc/zaptel.confspan=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15dchan=16bchan=17-31loadzone=itdefaultzone=it/etc/asterisk/zapata.conf[trunkgroups][channels]language=itcontext=from-zaptelsignalling=pri_netswitchtype=euroisdnrxwink=300 ; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines;;usedistinctiveringdetection=yesusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=no echotraining=800rxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nochannel = 1-15channel = 17-31---thats the debug when i try to call from the hipath Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1 -- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification)-- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null --byebivio
Re: [asterisk-users] detecting busy on queue transfer
does it solve the problem with j option?Do you have autofallthrough=yes in your general section of extensions.conf ?autofallthrough: New in 1.2. From the sample extensions.conf:
If autofallthrough is set, then if an extension runs out of things to
do, it will terminate the call with BUSY, CONGESTION or HANGUP
depending on Asterisk's best guess (strongly recommended). If
autofallthrough is not set, then if an extension runs out of things to
do, asterisk will wait for a new extension to be dialed (this is the
original behavior of Asterisk 1.0 and earlier).
On 10/2/06, lenz [EMAIL PROTECTED] wrote:
That's my fault in the example - I forgot to add in the j. Anyway whatis strange is that I get my dialplan to jump to position 108, but at thatpoint the agent is disconnected. I thought that when falling out of the
queuetransfer context, the control would be returned to the trasferer,after hearing the I'm sorry tone. Anything I'm missing here?l.In data Mon, 02 Oct 2006 00:36:30 +0200, Marco Mouta
[EMAIL PROTECTED] ha scritto: Hi, I've been looking the application dial on my asterisk server 1.2.9, and as far
CLI show application Dial j- Jump to priority n+101 if all of the requested channels were busy. It means that the application Dial on Asterisk 1.2 doesn't jump automatically on Busy to the extension n+101, only if you Dial it with
j argument!exten = _0.,7,Dial(Zap/g1/${EXTEN:1},,j)exten = _0.,108,NoOp(Got busy here) Or you should handle it on you priority 8 in your dialplan
exten = _0.,7,Dial(Zap/g1/${EXTEN:1}) exten = _0.,8,Goto(s-${DIALSTATUS},1) This is just an example. Hope it helps, please give me some feeback.
On 10/1/06, Lenz [EMAIL PROTECTED] wrote: I don't think that is the case - if I add a wait(10) after the step 108,
i.e. the busy detection, the agent seems to be disconnected immediately at the dial(), not after 10 seconds. That is what made me wonder what was going on. Yours
l. On Sun, 01 Oct 2006 17:49:15 +0200, Adam Goryachev [EMAIL PROTECTED]
wrote: Try adding this line: exten = _0.,109,Hangup Dunno if it will solve it, but might help :) Regards,
Adam -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] Siemens Hipath - asterisk, pri problem
please post your from-zaptel context in extensions.confOn 10/2/06, bivio [EMAIL PROTECTED] wrote: Hi all,i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones. asterisk (te110p) -- (TMS2) hipath 3550 in the future i'll connect the hipath to a telecom pri. the pri in the hipath is configured as EURO PP (with CRC4)i've already checked all previust post about this topic without any clue.many thanks/etc/zaptel.confspan=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15dchan=16bchan=17-31loadzone=itdefaultzone=it/etc/asterisk/zapata.conf[trunkgroups][channels]language=itcontext=from-zaptel signalling=pri_netswitchtype=euroisdnrxwink=300 ; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines;;usedistinctiveringdetection=yes usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=no echotraining=800rxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nochannel = 1-15channel = 17-31---thats the debug when i try to call from the hipath Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1 -- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification)-- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null --byebivio ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___
[asterisk-users] asterisk-oh323
I want to interconnect asterisk to a siemens HiQ20 which is configured as gatekeeker. The problem is that the HiQ20 does not accept gatekeeperrequests and sends immediately a reject with an undefinedReason. Is there a way to get asterisk-oh323 to skip this request? asterisk v1.2.12.1 asterisk-oh323 v0.7.3 both installed from aptitude Chris ___ --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] Siemens Hipath - asterisk, pri problem
Hello I connect HICOM to Asterisk Zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-isdn-external ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks signalling=pri_cpe switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown callerid=asreceived usedistinctiveringdetection=yes usecallingpres=yes ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel= 1-15,17-31 zaptel.conf # Global data span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bivio Sent: Monday, October 02, 2006 11:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Siemens Hipath - asterisk, pri problem Hi all, i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones. asterisk (te110p) -- (TMS2) hipath 3550 in the future i'll connect the hipath to a telecom pri. the pri in the hipath is configured as EURO PP (with CRC4) i've already checked all previust post about this topic without any clue. many thanks /etc/zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 dchan=16 bchan=17-31 loadzone=it defaultzone=it /etc/asterisk/zapata.conf [trunkgroups] [channels] language=it context=from-zaptel signalling=pri_net switchtype=euroisdn rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel = 1-15 channel = 17-31 --- thats the debug when i try to call from the hipath Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Making new call for cr 1 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (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 9f] 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: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02
[asterisk-users] suggest a configuration
I have to setup a pbx system for a company, can someone suggest a configuration. Currently their phone bill is 1600 a monthCurrenlty 27 phone lines1/2of the calls are long distanceI'd like the savings of a voip network, but also the reliability of a pstn/pri. How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] Siemens Hipath - asterisk, pri problem
2006/10/2, Marco Mouta [EMAIL PROTECTED]:
please post your from-zaptel context in extensions.confThanks to Giordano (immediate=yes) i see the first improvement now i hear the asterisk voice who says the number you digited is not in use, please check
now i'm tryng to understand how to andle the from-zaptel context, here it is[from-zaptel]exten = _X.,1,Set(DID=${EXTEN})exten = _X.,n,Goto(s,1)exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})exten = s,n,NoOp(DID is now ${DID})exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1); If there's no ext-did,s,1, that means there's not a no did/no cid route. Hangup.exten = s,n,Macro(hangup)exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})exten = s,n,Macro(from-zaptel-${CHAN},${DID},1); If nothing there, then treat it as a DIDexten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
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R: [asterisk-users] Siemens Hipath - asterisk, pri problem
Try to set
overlapdial=yes in your zapata, so thta whenu access to line ushould
have somethinghs of this
-- Starting simple switch on
'Zap/5-1' -- Accepting overlap voice call from '405' to
'unspecified' on channel 0/2, span 2
at this point u
can ear a continuos tone and input your dnid number (you will accept overlap
call from exten 100)
Ciao
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
bivioInviato: lunedì 2 ottobre 2006 10.58A: Asterisk
Users Mailing List - Non-Commercial DiscussionOggetto: Re:
[asterisk-users] Siemens Hipath - asterisk, pri
problem
2006/10/2, Marco Mouta [EMAIL PROTECTED]:
please
post your from-zaptel context in
extensions.confThanks to Giordano (immediate=yes) i
see the first improvement now i hear the asterisk voice who says "the number you
digited is not in use, please check" now i'm tryng to understand how to
andle the from-zaptel context, here it is[from-zaptel]exten =
_X.,1,Set(DID=${EXTEN})exten = _X.,n,Goto(s,1)exten =
s,1,NoOp(Entering from-zaptel with DID == ${DID}) ; If ($did == "") { $did =
"s"; }exten = s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})})exten
= s,n,NoOp(DID is now ${DID})exten =
s,n,GotoIf($["${CHANNEL:0:3}"="Zap"]?zapok:notzap) exten =
s,n(notzap),Goto(ext-did,${DID},1); If there's no ext-did,s,1, that means
there's not a no did/no cid route. Hangup.exten =
s,n,Macro(hangup)exten = s,n(zapok),NoOp(Is a Zaptel Channel) exten
= s,n,Set(CHAN=${CHANNEL:4})exten =
s,n,Set(CHAN=${CUT(CHAN,-,1)})exten =
s,n,Macro(from-zaptel-${CHAN},${DID},1); If nothing there, then treat it as
a DIDexten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN}) exten
= s,n,Goto(ext-did,${DID},1)
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Re: [asterisk-users] Siemens Hipath - asterisk, pri problem
try this:first: immediate=no ; otherwise what you r saying is to asterisk automatically dial when you hook up the phone!The problem is that in your context from-zaptel you are not dialing anywhere!i couldn't find you using any Dial(...)
That's why it doesn't work! try this:[from-zaptel]exten= _X.,1,Dial(Zap/G1/${EXTEN})exten= _X.,2,hangupUse it to dial a local extension, i suppose to dial out you are using a prefix
On 10/2/06, bivio [EMAIL PROTECTED] wrote:
2006/10/2, Marco Mouta [EMAIL PROTECTED]:
please post your from-zaptel context in extensions.confThanks to Giordano (immediate=yes) i see the first improvement now i hear the asterisk voice who says the number you digited is not in use, please check
now i'm tryng to understand how to andle the from-zaptel context, here it is[from-zaptel]exten = _X.,1,Set(DID=${EXTEN})exten = _X.,n,Goto(s,1)exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})exten = s,n,NoOp(DID is now ${DID})exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1); If there's no ext-did,s,1, that means there's not a no did/no cid route. Hangup.exten = s,n,Macro(hangup)exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})exten = s,n,Macro(from-zaptel-${CHAN},${DID},1); If nothing there, then treat it as a DIDexten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
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-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] Building the Perfect Box
We were also looking at this telecomproblem as well. A major complication,with regards to recovery planning,lies in the manner in which Local Loop Unbundling occurs. Even though communication companies may carry different logos, and profess to be independent orgs, they are all/mostly, invariably, in one form or another, dependent on the incumbent. In most, if not all cases, they share the incumbent's exchanges - whichusually are individually central tolarge areas.If a problem occurs in the Handover Distribution Frame (HDF)- this is where copper pairs are made available to CLECs in an exchange- non-discriminatory multihoming between providers would seem to be the best remedy. However, if the problem - as in the case of a major fire -is in the Main Distribution Frame (MDF) - where all the pre-unbundled wires are held-all providers, bar none,within that exchange will go down.Strategies such as Dual Parenting (connecting to two processors within the exchange), which is usually offered as a HA option, do little in such extreme cases -and whether you have one or two separate providers, under such circumstances,would matter little.However, there are other possibilities, such as multihoming with two different technologies - such as connecting to a Cable company, which, indeed, can provide you with an entirely different route - but, though there are ways around what I say next,you may be screwed on the DID issue. Even when the Cable companies Switching Center connects to the PSTN (SS7 ntwk), it does so with with very high availability as its aim- i.e., it is connectedto more than one, so if one, say, blows up, switching continues on another route - and the thing you then have to worry about is your building not catching fire, or being flooded in the middle of winter. You can also ask providers for information about their networks - with regards to location mapping+coverage. You may be - a big may be -lucky, and be in an area within an areawhere two or more exchangesdeliver services thereby allowing you to connect to two different centers. If both centers are run by the same incumbent, then potential DID issues are easily negated.Usually you can ask your providers for availability+performance figures (regional/local)-backdated for several years (they should have them on hand)- and mark such figures against your potential losses as tallied against particular frequencies relevant to your org.If you can do away with not having an SLA, it would usually indicate that your losses in relation to your revenue for the duration of, say, the fault, or, indeed, performance depreciation, are inconsequential. Some may say,in the case of business orgs,it could mean that they are not being assessed, or if they are, they are important enough not to let go. However, requirement usually preceeds the SLA, with the latter being mapped to an assessment of the former.You should alsobe very careful of what carriers tell you, as the information you gather is usually only as good as your point of reference. I.e., in many cases, their representative, who may or may not be keyed up, and may, or may not, be very polar with regards to the information she/he is prepared to/can deliver.This of course is an extremely interesting, as well as being a very wide and complex subject. And I do hope the above in some manner helps.Bayo"Jay R. Ashworth" [EMAIL PROTECTED] wrote: On Sat, Sep 30, 2006 at 11:14:08AM -0500, Brandon Galbraith wrote: On 9/30/06, Tim Panton [EMAIL PROTECTED] wrote: Just to amplify this point. I've tried to claim on an SLA. Our internet connection was down for a week due to a fire in BT's exchange. My provider refused to do anything (despite the premium SLA) on the basis that fires weren't covered. I switched providers to a cheaper one who didn't pretend to offer uptime :-) While an SLA is nice on paper, if your connection is business/mission-critical, always always always do redundancy yourself. Just having two connections from seperate providers is nice, multi-homing with two providers and having automatic failover is better (although this mostly applies to larger shops where having the phones/internet out per minute costs thousands of dollars/euros/etc.).But see also my earlier comments about the practical impossibility ofguaranteeing physical route diversity for multiple services in the sameformat (copper, fiber), even if from different putative carriers.This is *really* hard to make work; even the carriers themselves willoften *tell* you that you have PRD, and then you'll get backhoe fadedon all circuits anyway.Cheers,-- jra-- Jay R. Ashworth [EMAIL PROTECTED]Designer Baylink RFC 2100Ashworth Associates The Things I Think '87 e24St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274"That's women for you; you divorce them, and 10 years later,they stop having sex with you." -- Jennifer Crusie; _Fast_Women--Bandwidth and Colocation
Re: [asterisk-users] Siemens Hipath - asterisk, pri problem
whoa!! it works the fault was my ignorance of extension.confi modified the Marco advice in:[from-zaptel]exten= _X.,1,Dial(SIP/${EXTEN})exten= _X.,2,hangupso i can correctly call the astersik extension.
many many thanks allBivio
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Re: [asterisk-users] Where are the kernel sources?
Matthew Thompson wrote: yum install kernel-devel Should do the trick. It did, thanks. Jim. ___ --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] Can't get second line of Sangoma A200 to work.
I set up extension 120 for the first and 121 for the second. The first
one works as expected but I can't get a dial tone on the second one. I
hear a buzzing in the second port much like the first, but no dial
tone. I have power since the dtmf keys work OK. I tried changing the
exten = 121,hint,ZAP/2 to exten = 121,hint,ZAP/1 not really knowing
what the number after the ZAP/ was, but that didn't work.
Can someone give me a clue as to the proper configuration for the second
port? I also can't get the fxo ports to work either but I'll wait to
ask that one later after I've played with them a bit. I'm using
Tribox. Another strange thing is that when I make a change and press
the red bar at the top of the page, then Asterisk completely quits
working until I do a /etc/init.d/asterisk stop and start. But I can
live with that.
Thanks,
Jim.
extensions_additional.conf
[ext-local]
include = ext-local-custom
exten = 101,1,Macro(exten-vm,101,101)
exten = 101,hint,SIP/101
exten = ${VM_PREFIX}101,1,Macro(vm,101,DIRECTDIAL)
exten = 102,1,Macro(exten-vm,novm,102)
exten = 102,hint,SIP/102
exten = 120,1,Macro(exten-vm,novm,120)
exten = 120,hint,ZAP/1
exten = 121,1,Macro(exten-vm,novm,121)
exten = 121,hint,ZAP/2
zapata_additional.conf
;;[120]
signalling=fxo_ks
record_out=Adhoc
record_in=Adhoc
[EMAIL PROTECTED]
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=120 120
busydetect=no
busycount=7
accountcode=
channel=1
;;[121]
signalling=fxo_ks
record_out=Adhoc
record_in=Adhoc
[EMAIL PROTECTED]
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=121 121
busydetect=no
busycount=7
accountcode=
channel=2
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[asterisk-users] Call Quality / Echo / Problems
Hi all I'm having a problem getting usable quality from my Asterisk setup. *SETUP* 2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones in the house and 2 x FXO to receive calls and in the future faxes. Gentoo Linux Here is what I've done so far (1) Moved theTDM 400p (FXS, , FXO, FXO) to it's own interrupt (It was sharing in the past) cat /proc/interrupts CPU0 0: 10236724 XT-PIC timer 1:486 XT-PIC i8042 2: 0 XT-PIC cascade 5: 40694267 XT-PIC wctdm== 10: 196233 XT-PIC eth0 12:225 XT-PIC i8042 14: 247177 XT-PIC ide0 15: 26 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 (2) I'm running the latest kernel uname -r 2.6.17- (3) I am running the latest Asterisk Asterisk-1.2.12.1 Libpri-1.2.3 zaptel-1.2.8 I compiled Zaptel with make clean ; make linux26 ; make install (4) ztmonitor has become my friend the Ring sends the VU meter off the chart, but the voice is below half way. I have tried changing the rxgain, txgain but that doesn't improve much. It raises the Volume and I can heard better. But the feedback (What I hear of myself in my analog headset) is off tone , too loud and poor (5) ztspeed reports Count: 254114 (Not sure if that is good or bad) ??? (6) zttest reports --- Results after 474 passes --- Best: 100.00 -- Worst: 92.578125 -- Average: 99.713476 (7) I have even run ./fxotune which generated /etc/fxotune.conf 3=3,0,0,0,0,0,0,0,0 4=4,0,0,0,0,0,0,0,0 Found http://www.voip-info.org/wiki/view/Asterisk+fxotune But the -d -b 3 doesn't work only -i and -s are allowed. PROBLEM The Call tone has a tin can sound (too much highs and not enough lows, for those with musical backgrounds) The Volume has improved. It did sound like I was talking behind my hand in front of my mouth, but not anymore. The is a static HISSS that randomly comes and goes and gets so loud that it drowns out whatever the calling party is trying to say. It can be heard on both ends but very Loud on the FXS connected phone Any Ideas What I can try next Thanks All Barry ___ --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] can't transcode ilbc
I'm getting messages like 'WARNING[10263]: chan_sip.c:2552 sip_write: Asked to transmit frame type 8, while native formats is 1024 (read/write = 1024/1024)', where 8 = alaw and 1024 = ilbc. If I do show translation I get this: *CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 7 714 7 619 -93 - ulaw -17 - 1 9 2 114 -88 - alaw -17 1 - 9 2 114 -88 - g726 -23 8 8 - 8 720 -94 - adpcm -17 2 2 9 - 114 -88 - slin -16 1 1 8 1 -13 -87 - lpc10 -24 9 916 9 8 - -95 - g729 - - - - - - - - - - - speex -27121219121124 - - - ilbc - - - - - - - - - - - which I think means that there is no translation/transcoding path between ilbc and anything else. Is there some configuration option I need to set somewhere to allow this transcoding to take place or is there something about the ilbc protocol which makes transcoding a bad idea? The call in question is: SJphoneA - AsteriskA - mynetfone - AsteriskB - SJphoneB SJphoneA and SJphoneB both have a clear path to each other (tunnel), but when AsteriskA calls mynetfone, it uses nat. I don't think nat is a problem in this case though. Any suggestions? I'm also seeing the same thing when I try to make a call like: SJphone - Asterisk - PBX (via mISDN) SJphone is talking ilbc to Asterisk, but obviously Asterisk talks alaw to the PBX, and Asterisk refuses to transcode. James ___ --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] Spying a channel in a meetme
Hello,
I'm using the ChanSpy command for monitor a conversation of a channel which is
in a meetme conference. All comunications go throught voip, with some voip
phones attached to the lan and an external voip providor in order to make
external calls.
The problem is that sometimes the spy call can hear the other persons of the
conference, but sometimes it works ok. Almost all conferences are only of two
channels.
exten=s,1,Chanspy(${SPYCHAN}|q)
I will try using monitor mode in meetme application, but I prefer Chanspy
because I can spy the call always, not only when it is in a conference.
--
Eduard Martínez Bernal
Dpto. técnico
Barnatech, SCCL
Aragó nº 186 pral 4ª, 08011, Barcelona
+34 93 454 82 89
+34 93 454 58 69 (Fax)
E-mail: [EMAIL PROTECTED]
http://www.barnatech.com
--
Eduard Martínez Bernal
Dpto. técnico
Barnatech, SCCL
Aragó nº 186 pral 4ª, 08011, Barcelona
+34 93 454 82 89
+34 93 454 58 69 (Fax)
E-mail: [EMAIL PROTECTED]
http://www.barnatech.com
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[asterisk-users] attended transfer unreliable (2nd try)
Is there really nobody who has any idea about this?help would be really apreciated, as otherwise we're forced to buy a conventional pbxDate: 29.09.2006 15:33Subject: attended transfer unreliable To: asterisk-users@lists.digium.comHi,running asterisk 1.2.9 with freepbx 2.1.1, I have a strange problem:sometimes, call transfer works as expectet, and sometimes not. So far, I couldn't figure out any pattern in this behaviour, features.conf :featuredigittimeout = 1500atxfer = *3-works:# user enters *Sep 29 14:52:14 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:14 DEBUG[21578] channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:14 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18Sep 29 14:52:14 DEBUG[21578] res_features.c: Set time limit to 1500 Sep 29 14:52:14 VERBOSE[21578] logger.c: -- Attempting native bridge of SIP/210-859a and SIP/230-a983# user enters 3Sep 29 14:52:15 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:15 DEBUG[21578] channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:15 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18# here is the transfer: Sep 29 14:52:15 DEBUG[21578] res_features.c: Executing Attended Transfer SIP/210-859a, SIP/230-a983 (sense=2) XXXSep 29 14:52:15 VERBOSE[21578] logger.c: -- Started music on hold, class 'default', on SIP/210-859a -doesn't work:# user enters *Sep 29 09:17:54 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:54 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a Sep 29 09:17:54 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18Sep 29 09:17:54 DEBUG[20534] res_features.c: Set time limit to 1500Sep 29 09:17:54 VERBOSE[20534] logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a#user enters 3Sep 29 09:17:55 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:55 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a Sep 29 09:17:55 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18# no transferSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED]' of Request 102: Match FoundSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED] ' of Request 103: Match FoundSep 29 09:17:55 VERBOSE[20534] logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a--when we have a timeout, it looks different: Sep 29 12:00:34 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:34 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746Sep 29 12:00:34 DEBUG[21122] res_features.c: Feature interpret: chan=Zap/2-1, peer=SIP/240-6746, sense=2, features=18 Sep 29 12:00:34 DEBUG[21122] res_features.c: Set time limit to 1500Sep 29 12:00:36 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:36 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746 Sep 29 12:00:36 DEBUG[21122] res_features.c: Timed out for feature!hope your can help meStefan ___ --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] attended transfer unreliable (2nd try)
Stefan Friedrich wrote: Is there really nobody who has any idea about this? help would be really apreciated, as otherwise we're forced to buy a conventional pbx Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN? There have been a few fixes in the branch that may help. You can get instructions towards the center of the page at: http://www.asterisk.org/download Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Dial and connect to sip provider works, but no audio.
This is strange. I upgraded from an older [EMAIL PROTECTED] that was working to the latest Tribox. I also added a A204 board, but for some reason neither the Grandstream phone or a phone connected to the Linksys ATA has any audio either way via the Telasip connection. Audio works OK between the phones, so I'm pretty sure the extension configuration is OK.. Here's my sip configs. I added the [from-pstn] to this file because I didn't see it defined anywhere else. I realize it will go away when I change the extensions but it wasn't working so I thought I'd try it. I don't see much difference in configuration from when it worked and now, other than the missing [from-pstn] block. Thanks for any help. Jim. sip_additional.conf [EMAIL PROTECTED] [101] username=101 type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=101 101 [102] username=102 type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=102 102 [from-pstn] type=user qualify=yes insecure=very host=xxx.telasip.com [telasip] username=xxx type=friend secret=xxx.yyy qualify=yes insecure=very host=xxx.telasip.com fromuser=xxx fromdomain=xxx.telasip.com dtmfmode=rfc2833 context=from-pstn ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding nat=1 to each peer definition to ; solve translation problems. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying The number you have dialed is not in service. Please check the ; number and try again. context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf ~ ___ --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] Issues with calling certain phone numbers...
Hello,
I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbers
also via SIP through an AS5350 which has an E1 ISN PRI attached.
I have a PSTN operator number (say 012345678) routed to three SIP
extensions (01,21,20) and numbers to directly reach extensions from
outside (say 98765432XX, where 00 XX 99).
[outsidetoinside]
exten = 012345678,1,Dial(SIP/01,10,t);
exten = 012345678,n,Dial(SIP/21,10,t);
exten = 012345678,n,Dial(SIP/20,10,t);
exten = 012345678,n,Goto(1);
exten = _98765432XX,1,Dial(SIP/${EXTEN:8},60);
exten = _98765432XX,n,Hangup();
All two digits numbers dialed from extensions are routed to other
extensions, three digit numbers get routed to the PSTN Gateway.
[insidetooutside]
exten = 012345678,1,Dial(SIP/01);
exten = 012345678,n,Hangup();
exten = _98765432XX,1,Dial(SIP/${EXTEN:8});
exten = _98765432XX,n,Hangup();
exten = _XX.,1,Set(CALLERID(number)=012345678);
exten = _XX.,n,Dial(SIP/[EMAIL PROTECTED]);
exten = _XX.,n,Hangup();
exten = _XX,1,Dial(SIP/${EXTEN},30,t);
exten = _XX,n,Hangup();
The problem is that three digit numbers like 187 (which is a public
reachable PSTN number in my country, so I can reach it via the E1) is
not actually routed to the PSTN gateway (as it should).
I tried debugging SIP and see no request made to the AS5350. Is there a
command in the asterisk cli to debug how dialplan logic matches
requests? What could be crong?
TIA
Luca
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Re: [asterisk-users] attended transfer unreliable (2nd try)
Doug Lytle wrote: Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN? Transfer (rather, dynamic features in general) is broken in 1.2.12.1: http://bugs.digium.com/view.php?id=7982 So you should try the version from the SVN branch. Yours, Florian. ___ --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 queues with SER, aka sip show peers
Hi,I am trying to integrate asterisk queues with SER.We have our queues set up in the following manner:An entry in the queue members table consts of the queue name and a SIP address.For example,queuename | member -support-q | SIP/5558675309We have observed the following behavior:If the first phone is registered directly to asterisk, then the queue knows about the phone, and puts it in the queue rotation. I would like to get this same functionality using SER as the registration server.Currently, if a phone is registered with the SER proxy, * knows nothing about it until the phone calls an extension on the asterisk server. After that, the phone shows up in sip show peers, but with address (unknown).In neither case is the queue aware of the phone.Here is the setup:Asterisk is set up in general to be real time, including sip peers and sip users. in sip.conf, [general]... rtcachefriends=yes register = [EMAIL PROTECTED]: [EMAIL PROTECTED]/2342342345[ser]type=friend insecure=veryhost=sip.proxy.comcanreinvite=yesautocreatepeer=yescontext=proxyAsterisk is successfully registering with SER, and sip show peers shows the SER box as a peer, although I think that is irrelevant of if the SER service is running or not. What do I need to do to get Asterisk to be aware of the clients?Mark Price ___ --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] 480i phone: Is there a trick to registering with *?? --Solved
1. Reset to factory defaults
2. Put registration information under Global SIP and not line 1
3. Put in IP address of Asterisk server in every field that says Proxy
4. THE TRICK: Phone number field in Global SIP must have account name, not
actual phone number.
Working awesome so far, thanks Dave
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 30, 2006 3:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] 480i phone: Is there a trick to
registering with *??
oj that's awesome info thanks i will try it when i get back to the office on
monday (hey even geeks have to ride bikes on the weekend esp with the leaves
turning) fwiw i used the web interface and it's as brutal as a Grandstream
but other than that this little number looks *so* sweet - cordless range
seems *very* good
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 30, 2006 1:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 480i phone: Is there a trick to
registering with *??
On Sat, 2006-09-30 at 00:47 -0600, Colin Anderson wrote:
yeah, weird that, i did set the proxy after I posted so it's now
sip:[EMAIL PROTECTED] but still no dice. There is a port number as
well
that I left at 0, should I change it to 5060?
btw, the hardware is a sweet little package if I get this working I could
see this being my favorite phone
How are you setting up the phone, by the web interface, the on screen or
from tftp?
Don't mix them the phones get confused. If you have do a full factory
reset to get it's full attention.
I only use tftp and they run as sweet as a nut.
my dhcpd.conf hands out the ip and filename of the main config file
host Aastra480i {
hardware ethernet 00:08:5D:18:35:52;
fixed-address 192.168.1.5;
filename /aastra.cfg;
}
The file aastra.cfg contains
dhcp: 1
tftp server: 192.168.1.253
subnet mask: 255.255.255.0
default gateway: 192.168.1.1
dns1: 192.168.1.253
time server disabled: 0
time server1: 192.168.1.253
time format: 1
date format: 0
language: 0
tone set: France
time zone name: FR-Paris
time zone code: CET
time zone minutes: 60
sip nortel nat support: 0
sip proxy ip: 192.168.1.253
sip proxy port: 5060
sip registrar ip: 192.168.1.253
sip registrar port: 5060
sip digit time out: 3
sip registration period: 60
sip session timer: 0
sip rtp port: 8000
sip transport protocol: 1
sip use basic codecs: 1
sip dial plan: X+#
sip dial plan terminator: 0
sip mode: 0
sip vmail: *10
sip blf subscription period: 60
sip silence suppression: 0
sip update callerid: 1
sip allow auto answer: 1
sip intercom mute mic: 1
sip intercom type: 2
sip intercom prefix code: *55
sip intercom line: 1
directed call pickup: 1
call waiting tone: 1
directory 1: company.csv
auto resync mode: 3
auto resync time: 01:30
priority alerting enabled: 1
map conf key to: 2663
stuttered disabled: 0
--
and 00085D183552.cfg (not uppercase) contains
#
#
# 480i
#
#
sip screen name: Linux Autrement
sip line1 auth name: 2001
sip line1 password: password
sip line1 user name: 2001
sip line1 display name: Dave Cotton
sip line1 screen name: Dave Cotton
sip line1 vmail: *10#
sip line2 auth name: 2001
sip line2 password: password
sip line2 user name: 2001
sip line2 display name: Dave Cotton
sip line2 screen name: Dave Cotton
sip line2 vmail: *10#
sip line3 auth name: 2001
sip line3 password: password
sip line3 user name: 2001
sip line3 display name: Dave Cotton
sip line3 screen name: Dave Cotton
sip line3 vmail: *10#
sip line4 auth name: 2001
sip line4 password: password
sip line4 user name: 2001
sip line4 display name: Dave Cotton
sip line4 screen name: Dave Cotton
sip line4 vmail: *10#
sip explicit mwi subscription: 1
sip intercom type: 2
sip intercom prefix code: *55
sip intercom line: 1
#blf
softkey1 type: blf
softkey1 label: Elaine
softkey1 value: 2002
softkey1 states: idle connected incoming outgoing
softkey1 line: 1
#speeddial
softkey8 type: speeddial
softkey8 label: Paris
softkey8 value: 5307
softkey8 states: idle connected incoming outgoing
softkey8 line: 1
sip.conf is then
[2001]
type = peer
secret = password
host = dynamic
defaultip = 192.168.1.5
mailbox = 2001
language = en
subscribecontext = internal
context = internal
canreinvite = no
dtfmmode = auto
disallow = all
allow = ulaw
qualify = no; Aastras don't seem to answer qualify so set as no
callgroup = 1
pickupgroup = 1
call-limit = 4
callerid = Dave Cotton 2001
Hope this helps.
--
Dave Cotton [EMAIL PROTECTED]
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Re: [asterisk-users] Issues with calling certain phone numbers...
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06, Luca Corti
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbersalso via SIP through an AS5350 which has an E1 ISN PRI attached.I have a PSTN operator number (say 012345678) routed to three SIPextensions (01,21,20) and numbers to directly reach extensions from
outside (say 98765432XX, where 00 XX 99).[outsidetoinside]exten = 012345678,1,Dial(SIP/01,10,t);exten = 012345678,n,Dial(SIP/21,10,t);exten = 012345678,n,Dial(SIP/20,10,t);
exten = 012345678,n,Goto(1);exten = _98765432XX,1,Dial(SIP/${EXTEN:8},60);exten = _98765432XX,n,Hangup();All two digits numbers dialed from extensions are routed to otherextensions, three digit numbers get routed to the PSTN Gateway.
[insidetooutside]exten = 012345678,1,Dial(SIP/01);exten = 012345678,n,Hangup();exten = _98765432XX,1,Dial(SIP/${EXTEN:8});exten = _98765432XX,n,Hangup();exten = _XX.,1,Set(CALLERID(number)=012345678);
exten = _XX.,n,Dial(SIP/[EMAIL PROTECTED]);exten = _XX.,n,Hangup();exten = _XX,1,Dial(SIP/${EXTEN},30,t);exten = _XX,n,Hangup();The problem is that three digit numbers like 187 (which is a public
reachable PSTN number in my country, so I can reach it via the E1) isnot actually routed to the PSTN gateway (as it should).I tried debugging SIP and see no request made to the AS5350. Is there acommand in the asterisk cli to debug how dialplan logic matches
requests? What could be crong?TIALuca___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] Issues with calling certain phone numbers...
My mistake sorry for last postOn 10/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06,
Luca Corti
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbersalso via SIP through an AS5350 which has an E1 ISN PRI attached.I have a PSTN operator number (say 012345678) routed to three SIP
extensions (01,21,20) and numbers to directly reach extensions from
outside (say 98765432XX, where 00 XX 99).[outsidetoinside]exten = 012345678,1,Dial(SIP/01,10,t);exten = 012345678,n,Dial(SIP/21,10,t);exten = 012345678,n,Dial(SIP/20,10,t);
exten = 012345678,n,Goto(1);exten = _98765432XX,1,Dial(SIP/${EXTEN:8},60);exten = _98765432XX,n,Hangup();All two digits numbers dialed from extensions are routed to otherextensions, three digit numbers get routed to the PSTN Gateway.
[insidetooutside]exten = 012345678,1,Dial(SIP/01);exten = 012345678,n,Hangup();exten = _98765432XX,1,Dial(SIP/${EXTEN:8});exten = _98765432XX,n,Hangup();exten = _XX.,1,Set(CALLERID(number)=012345678);
exten = _XX.,n,Dial(SIP/[EMAIL PROTECTED]);exten = _XX.,n,Hangup();exten = _XX,1,Dial(SIP/${EXTEN},30,t);exten = _XX,n,Hangup();The problem is that three digit numbers like 187 (which is a public
reachable PSTN number in my country, so I can reach it via the E1) isnot actually routed to the PSTN gateway (as it should).I tried debugging SIP and see no request made to the AS5350. Is there acommand in the asterisk cli to debug how dialplan logic matches
requests? What could be crong?TIALuca___--Bandwidth and Colocation provided by
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-- Com os melhores cumprimentos,Marco Mouta
-- Com os melhores cumprimentos,Marco Mouta
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RE: [asterisk-users] 480i phone: Is there a trick to registering with *?? --Solved, first impressions
First impressions:
1. Audio is decent on the cordless, not as good as a Panasonic but quite
usable.
2. Cordless range is awesome. Went 150 metres away, 2 buildings over,
through probably a dozen walls. No problem.
3. Possible for handset to independently originate and terminate calls while
base station is doing the same.
4. MWI works on both handset and base station, first time, correctly. VM
count is incremented and deincremented correctly.
5. Caller ID name + number is rec'd and sent correctly
6. Line, Xfer and Conf keys work exactly as expected.
7. Speakerphone is good.
8. Screen and menuing interface is precisely the same as a Vista 350 - good
for bonehead users
9. A nice touch is that the cordless handset base station operates without
being plugged into the base station - it's wireless-to-wireless
yep, it's official. this is now my favorite Asterisk phone. My only niggle
is to clean up the web provisioning interface it's super sparse and maybe a
little better codec selection.
good job, Aastra! -Users: check this phone out it is a good value.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Monday, October 02, 2006 9:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] 480i phone: Is there a trick to
registering with *?? --Solved
1. Reset to factory defaults
2. Put registration information under Global SIP and not line 1
3. Put in IP address of Asterisk server in every field that says Proxy
4. THE TRICK: Phone number field in Global SIP must have account name, not
actual phone number.
Working awesome so far, thanks Dave
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 30, 2006 3:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] 480i phone: Is there a trick to
registering with *??
oj that's awesome info thanks i will try it when i get back to the office on
monday (hey even geeks have to ride bikes on the weekend esp with the leaves
turning) fwiw i used the web interface and it's as brutal as a Grandstream
but other than that this little number looks *so* sweet - cordless range
seems *very* good
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 30, 2006 1:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 480i phone: Is there a trick to
registering with *??
On Sat, 2006-09-30 at 00:47 -0600, Colin Anderson wrote:
yeah, weird that, i did set the proxy after I posted so it's now
sip:[EMAIL PROTECTED] but still no dice. There is a port number as
well
that I left at 0, should I change it to 5060?
btw, the hardware is a sweet little package if I get this working I could
see this being my favorite phone
How are you setting up the phone, by the web interface, the on screen or
from tftp?
Don't mix them the phones get confused. If you have do a full factory
reset to get it's full attention.
I only use tftp and they run as sweet as a nut.
my dhcpd.conf hands out the ip and filename of the main config file
host Aastra480i {
hardware ethernet 00:08:5D:18:35:52;
fixed-address 192.168.1.5;
filename /aastra.cfg;
}
The file aastra.cfg contains
dhcp: 1
tftp server: 192.168.1.253
subnet mask: 255.255.255.0
default gateway: 192.168.1.1
dns1: 192.168.1.253
time server disabled: 0
time server1: 192.168.1.253
time format: 1
date format: 0
language: 0
tone set: France
time zone name: FR-Paris
time zone code: CET
time zone minutes: 60
sip nortel nat support: 0
sip proxy ip: 192.168.1.253
sip proxy port: 5060
sip registrar ip: 192.168.1.253
sip registrar port: 5060
sip digit time out: 3
sip registration period: 60
sip session timer: 0
sip rtp port: 8000
sip transport protocol: 1
sip use basic codecs: 1
sip dial plan: X+#
sip dial plan terminator: 0
sip mode: 0
sip vmail: *10
sip blf subscription period: 60
sip silence suppression: 0
sip update callerid: 1
sip allow auto answer: 1
sip intercom mute mic: 1
sip intercom type: 2
sip intercom prefix code: *55
sip intercom line: 1
directed call pickup: 1
call waiting tone: 1
directory 1: company.csv
auto resync mode: 3
auto resync time: 01:30
priority alerting enabled: 1
map conf key to: 2663
stuttered disabled: 0
--
and 00085D183552.cfg (not uppercase) contains
#
#
# 480i
#
#
sip screen name: Linux Autrement
sip line1 auth name: 2001
sip line1 password: password
sip line1 user name: 2001
sip line1 display name: Dave Cotton
sip line1 screen name: Dave Cotton
sip line1 vmail: *10#
sip line2 auth name: 2001
sip line2 password: password
sip line2 user name: 2001
sip line2 display name: Dave Cotton
sip line2 screen name: Dave Cotton
sip line2 vmail: *10#
sip line3 auth name: 2001
sip line3 password: password
sip
[asterisk-users] Asterisk 1.2.10 and SCCP
I've got an interesting situation where I am running Asterisk 1.2.10 with
the chan_sccp2 implementation. The system crashes periodically, and each
time I get similar looking results when using gdb on the core files.
It looks almost as if someone is transferring a call to someone's voicemail
and the transferree decides not to leave a message and hangs up. Anyone
better able to make sense of the gdb bt full's and maybe point me in a
direction for resolution?
Unfortunately, I have to utilize SCCP as we are using 7960's and need
presence support which is impossible with the SIP image.
I would consider moving to Asterisk 1.4 if the skinny support where any
better, but I don't have any evidence that it is. Can anyone comment on the
native skinny support in 1.4?
My gdp bt full's are here. Thanks.
First Core-
(gdb) bt full
#0 0xb7f8f367 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x08060cd8 in ast_queue_hangup (chan=0x0) at lock.h:606
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0, mallocd
= 0, offset = 0, src = 0x0, data = 0x0, delivery = {
tv_sec = 0, tv_usec = 0}, prev = 0x0, next = 0x0}
#2 0xb724e0a9 in sccp_channel_transfer_complete (c=0xb5d05690) at
sccp_channel.c:861
transferred = (struct ast_channel *) 0x81faeb8
transferee = (struct ast_channel *) 0xb5d10ae0
destination = (struct ast_channel *) 0x0
peer = (sccp_channel_t *) 0x8223bf8
d = (sccp_device_t *) 0x81c75e0
__PRETTY_FUNCTION__ = sccp_channel_transfer_complete
#3 0xb724d911 in sccp_channel_transfer (c=0xb5d05690) at sccp_channel.c:731
d = (sccp_device_t *) 0x81c75e0
newcall = (sccp_channel_t *) 0x81c75e0
__PRETTY_FUNCTION__ = sccp_channel_transfer
#4 0xb72575a5 in sccp_sk_transfer (d=0x81c75e0, l=0x81d1100, c=0x0) at
sccp_softkeys.c:86
No locals.
#5 0xb724acd8 in sccp_handle_soft_key_event (s=0x8153cc8, r=0x0) at
sccp_actions.c:1194
d = (sccp_device_t *) 0x81c75e0
c = (sccp_channel_t *) 0xb5d05690
l = (sccp_line_t *) 0x81d1100
k = (sccp_speed_t *) 0x0
event = 4
line = 1
callid = 449
__PRETTY_FUNCTION__ = sccp_handle_soft_key_event
#6 0xb72426be in sccp_handle_message (r=0xb5d10090, s=0x8153cc8) at
chan_sccp.c:583
mid = 38
__PRETTY_FUNCTION__ = sccp_handle_message
#7 0xb7258be7 in sccp_socket_thread (ignore=0x0) at sccp_socket.c:304
fset = {fds_bits = {0, 262144, 0 repeats 30 times}}
res = 0
maxfd = 92
now = 1159467448
s = (sccp_session_t *) 0x8153cc8
s1 = (sccp_session_t *) 0x0
m = (sccp_moo_t *) 0x0
tv = {tv_sec = 0, tv_usec = 32000}
sigs = {__val = {138432515, 0 repeats 31 times}}
attr = {__detachstate = 0, __schedpolicy = 0, __schedparam =
{__sched_priority = 1}, __inheritsched = 4096, __scope = 0,
__guardsize = 0, __stackaddr_set = 0, __stackaddr = 0x0, __stacksize = 0}
__PRETTY_FUNCTION__ = sccp_socket_thread
#8 0xb7f8d0fb in start_thread () from /lib/tls/libpthread.so.0
No symbol table info available.
#9 0xb7e7c99e in clone () from /lib/tls/libc.so.6
No symbol table info available.
---Second Core
(gdb) bt full
#0 0xb7f1d367 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x08060cd8 in ast_queue_hangup (chan=0x0) at lock.h:606
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0, mallocd
= 0, offset = 0, src = 0x0, data = 0x0, delivery = {
tv_sec = 0, tv_usec = 0}, prev = 0x0, next = 0x0}
#2 0xb71dc0a9 in sccp_channel_transfer_complete (c=0xb60059d0) at
sccp_channel.c:861
transferred = (struct ast_channel *) 0x82109f0
transferee = (struct ast_channel *) 0xb6001710
destination = (struct ast_channel *) 0x0
peer = (sccp_channel_t *) 0x82090d0
d = (sccp_device_t *) 0x81c3fa8
__PRETTY_FUNCTION__ = sccp_channel_transfer_complete
#3 0xb71db911 in sccp_channel_transfer (c=0xb60059d0) at sccp_channel.c:731
d = (sccp_device_t *) 0x81c3fa8
newcall = (sccp_channel_t *) 0x81c3fa8
__PRETTY_FUNCTION__ = sccp_channel_transfer
#4 0xb71e55a5 in sccp_sk_transfer (d=0x81c3fa8, l=0x81cd9a8, c=0x0) at
sccp_softkeys.c:86
No locals.
#5 0xb71d8cd8 in sccp_handle_soft_key_event (s=0x81e1428, r=0x0) at
sccp_actions.c:1194
d = (sccp_device_t *) 0x81c3fa8
c = (sccp_channel_t *) 0xb60059d0
l = (sccp_line_t *) 0x81cd9a8
k = (sccp_speed_t *) 0x0
event = 4
line = 1
callid = 391
__PRETTY_FUNCTION__ = sccp_handle_soft_key_event
#6 0xb71d06be in sccp_handle_message (r=0xb6000470, s=0x81e1428) at
chan_sccp.c:583
mid = 38
__PRETTY_FUNCTION__ = sccp_handle_message
#7 0xb71e6be7 in sccp_socket_thread (ignore=0x0) at sccp_socket.c:304
fset =
Re: [asterisk-users] Fax detection ...
On Mon, Oct 02, 2006 at 10:43:44AM +0800, Steve Underwood wrote: Jay R. Ashworth wrote: On Sun, Oct 01, 2006 at 02:58:37PM -0700, Lee Howard wrote: Well, fax detection isn't entirely reliable anyway. Even if you assume that your fax detection feature and operation is flawless in properly detecting fax tones (and that most likely would be a specious assumption), not all calling fax machines send fax tones. So, y'know, that assertion gets made a lot. What's the turn rate of fax machines in the market? 3 years? 5? CNG tones are *well* over 10 years old, no? What relevance does that have to CNG? It was a feature of the original spec 30 years ago. Well, perhaps I wasn't paying attention, but I thought that CNG tones *had as their purpose* making receive FAX detection trivial. That would tend to make the question on-point, would it not? What percentage of fax calls are sent without CNG tones these days? Quite a lot. A large number of FAX machines have CNG turned off. On many machines, if select features like sharing a line between FAX and answering machine CNG, CED and various other useful behaviour might be disabled. My personal experience is that I've never seen a consumer-grade fax machine with send-CNG turned off, and I don't *think* I've ever seen one on which there was a knob *to* turn it off; I would be less sure about fax modems -- those may have a knob, but I would expect it to default on. Could you expand on what behaviour you think CNG breaks? Cause I'm not modeling it, mentally... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] G726 prompts
On Mon, Oct 02, 2006 at 01:33:43PM +1000, RR wrote: does anyone happen to know of a good utility or CLI tool to convert prompts into a g.726 format? I tried using the convert utility in (*) but it doens't like G.726. I understand I can just hunt around the net for it, but if someone knows one off-hand that I can run on linux and even run it inside a script that would be great. It seems unreasonably difficult to get a list of the supported formats, but does sox (http://sox.sourceforge.net/) do what you need? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] recommended application for salesman using asterisk
On 9/30/06, Tim Panton [EMAIL PROTECTED] wrote: On 29 Sep 2006, at 19:20, Yu Safin wrote: Hi, I am a salesman currently using asterisk to contact my customers. So far, I have asterisk connected to two PSTN analog lines where I only receive phones calls. Then, I have asterisk connected to a VoIP service company for terminating my phone calls. I also kept one PTSN phone line to place calls to my cellular when I am on the road. This is done by giving the caller an option to find me on my cellular. I started to tinker with my PC and I can now receive and place calls using a soft-phone (iaxclient). Then I started to wonder if there is an application that will allow me to use a friendly WEB interface along side with my soft-phone to quickly place phone calls from an address book and when I call arrives, to bring up information about the caller if present in my address book. I also conceived the idea that I might not even have the imagination that some members of this list may have in terms of how else I can be exploiting asterisk e.g. callback from messages left. Another one is a more sophisticated find-me service that I can manage from a WEB interface. I am open to ideas and suggestions. Please show me where to go. Remeber, if I spend too much time as an engineer I go hungry. I need to be in front of people selling my company services and not tinkering with applications. Yep, you could replace iaxclient with a web-embedded softphone like Corraleta SDK (ours) or Mozphone. (There are also a couple of SIP ones, but you are using IAX already, so stick with it) Then when you click on a number in the web page it would call that number for you. You'd need to do a bit of javascript coding to tweak the look and feel of Corraleta, but mostly people can get it the way they want pretty quickly. Tim Panton www.mexuar.com ___ --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 Tim I am at the Web site exploring the product. Thank you. ___ --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] Fax detection ...
Jay R. Ashworth wrote: My personal experience is that I've never seen a consumer-grade fax machine with send-CNG turned off, and I don't *think* I've ever seen one on which there was a knob *to* turn it off; I would be less sure about fax modems -- those may have a knob, but I would expect it to default on. On fax modems the way to silent-dial (and I believe that this was a norm from early-on) to to add an @ at the end of the dialstring: ATDT5551212@. I would be very surprised to find any modern fax modem that does not have this capability. I don't know of any specific fax machine that has such a knob to turn CNG off. But my contention wasn't that it was consumer-grade fax machines that were the main culprit here, but rather fax servers (PCs with fax modems in them). And depending on what industry you are sampling, those may actually consitute a fair amount of the caller pool. (For example, some industry software - like insurance agent application software - will have built-in fax features that will use the PC's fax modem - and the application vendor may insist on that feature being used.) I cannot cite specific software that does it, but I suspect that most fax application developers are aware of the ability to silent-dial, and the reasons why it may be employed. As I've said before, one reason, as an example, is to make the modem capable of hearing ringback - so that it knows if the call has been answered or not (which itself is a unreliable endeavor). Another reason is to avoid annoying the receiver moreso on a call to a wrong-number. It's off the topic of silent dialing but on the topic of this thread... Brother fax machine manuals state that it is possible for them to erroniously detect certain voices or music as CNG tones, and if that becomes a problem to disable fax detection. And that's basically another point along the lines that say that fax detection is not completely reliable. Lee. ___ --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] t1 voip to failover pri
I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to havea PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only?is anyone out there, using a VOIP only with no failover? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] t1 voip to failover pri
stan ford wrote: I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to have a PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only? is anyone out there, using a VOIP only with no failover? We're using VOIP only, no failover. Furthermore we're using it on a cable internet connection. We have a cheap dsl connection for backups. It's been up for about 2 months now and has only been out twice for a small period of time. When that happens the DSL takes over. I don't pretend that this is in anyways comparable to PSTN service but it works pretty well for us. We have three locations. Two of which are set up the same way, the third just has 3 stations and just registers with one of the asterisk boxes at the other locations. I think when you're talking enterprise you definitely want to go with a t1 or two t1's for backup. (I don't really understand how a PRI gives you more reliability than a T circuit. They run over the same copper don't they??) For our purposes however (and I'd like to think I speak for a lot of mid size businesses with 50 employees) our setup works wonderful. It costs us about $600 all in all (internet access + VOIP) and that's a FAR cry from what we were paying through Covad before. Of course there always will be exceptions (People that need 100% guaranteed uptime), but for the size of our business this works. The only part that REALLY concerns me is our DID's. If our DID provider ever goes down we are screwed. Anyone know of any failsafes for 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
[asterisk-users] !! No channel map, no channel, and no ds1? What am I supposed to identify?
Hi guys,
I'm tryng to estabilish a trunk with an Alcatel 4200 Pabx. In asterisk i'm using a Digium TE110P as E1.
The span is ok with green led, but when pabx make calls to asterisk, i received this error:
asterisk*CLI
!! Unexpected Channel selection 3
-- Accepting call from '3069' to '30818559' on channel 1/31, span 1
-- Executing Dial(Zap/31-1, SIP/[EMAIL PROTECTED]|20|Tt) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/fp-33133000-09fdfa90 is ringing
!! Unexpected Channel selection 3
-- SIP/fp-33133000-09fdfa90 answered Zap/31-1
!! No channel map, no channel, and no ds1? What am I supposed to identify?
!! Unable to add IE 'Channel Identification'
== Spawn extension (default, 30818559, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
Sep 23 20:13:25 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
Sep 23 20:13:29 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
My configuration files is:
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
trunkgroup = 1,16
spanmap = 1,1,1
language=uk
context=default
switchtype=euroisdn
signalling=pri_net
group=1
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes
channel = 1-15,17-31
/etc/asterisk/extensions.conf
# SIP - Alcatel
exten= 331330XX,1,Dial(Zap/g1/${EXTEN})
exten= 331330XX,2,Hangup
# Alcatel - SIP
exten= _,1,Dial(SIP/[EMAIL PROTECTED],20,Tt) #
exten= _,2,Hangup
What can be hrong in this configuration ???
Thanks.
-- Frederico Madeira[EMAIL PROTECTED]
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RE: [asterisk-users] t1 voip to failover pri
If reliability is the issue, then use the PRI *first* then failover to VoIP. If cost savings are the issue, use VoIP then have a 2nd VoIP provider to fail over to, and no PRI. In either scenario, inbound call routing is thorny, some guys that provide both PRI and VoIP can route calls automatically on failover. However, you *will* get 5 9's in any kind of PRI scenario, that is what it is designed to do. If your voicedowntime is measured in hundreds or thousands of dollars a minute, use a PRI. ROI on Asterisk depends on how you look at it. I enjoy the fact that licensing costs are zero and you can make it do way cool stuff, and you have access to a huge 3rd party market. But I would get my ass canned if I went to VoIP only and it went down. I do have a hybrid install with a SIP 12 channel connection and 2 BRI's for failover, and the cost savings are 30% over a frac PRI. You don't nessisarily have to do a 1-1 backup of your voice channels, all you need is enough to support 80% or so of your estimated averageconcurrent use and chances are your users will never know the difference in a failover situation unless it's another 9/11 and everyone is calling out. In that case, you can say, hey, it's another 9/11, no kidding the phones didn't work. -Original Message-From: stan ford [mailto:[EMAIL PROTECTED]Sent: Monday, October 02, 2006 10:55 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] t1 voip to failover pri I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to havea PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only? is anyone out there, using a VOIP only with no failover? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Conversations Mix
Hello, I have a problem with an adit 600 and a T400P card. This equipment was in a shelf for 2 years and when we connected an install it asterisk everything worked fine. But then we started receiving complaints that a person pick up their phone and will hear some other conversation. It happens with internal calls and outbound calls too. The channel bank has 4 fxs cards and one fxo card. From the console it seems that for some phones asterisk does not register the hook off nor the dialing. Any hints how to solve this ? regards,Iván ___ --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] 480i phone: Is there a trick to registering with *??
I set up mine with the web interface but I notice that some settings can only be made by config files. Do you know how to extract the current config file from the phone? Here's how I set up the web interface: Authentication Name: aastra480_1 Password: password BLA Number: blank Line Mode: Generic Proxy Server: 192.168.0.80 Proxy Port: 5060 Outbound Proxy Server: 192.168.0.80 Outbound Proxy Port: 5060 Registrar Server: 192.168.0.80 Registrar Port: 5060 Registration Period: 300 Dave Cotton wrote: On Sat, 2006-09-30 at 09:35 +0200, Dave Cotton wrote: and 00085D183552.cfg (not uppercase) contains Whoops note ___ --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] Re: can't transcode ilbc
On 2006-10-02 04:02:56 -0700, James Harper [EMAIL PROTECTED] said: I'm getting messages like 'WARNING[10263]: chan_sip.c:2552 sip_write: Asked to transmit frame type 8, while native formats is 1024 (read/write = 1024/1024)', where 8 = alaw and 1024 = ilbc. If I do show translation I get this: *CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 7 714 7 619 -93 - ulaw -17 - 1 9 2 114 -88 - alaw -17 1 - 9 2 114 -88 - g726 -23 8 8 - 8 720 -94 - adpcm -17 2 2 9 - 114 -88 - slin -16 1 1 8 1 -13 -87 - lpc10 -24 9 916 9 8 - -95 - g729 - - - - - - - - - - - speex -27121219121124 - - - ilbc - - - - - - - - - - - which I think means that there is no translation/transcoding path between ilbc and anything else. Is there some configuration option I need to set somewhere to allow this transcoding to take place or is there something about the ilbc protocol which makes transcoding a bad idea? The call in question is: SJphoneA - AsteriskA - mynetfone - AsteriskB - SJphoneB SJphoneA and SJphoneB both have a clear path to each other (tunnel), but when AsteriskA calls mynetfone, it uses nat. I don't think nat is a problem in this case though. Any suggestions? I'm also seeing the same thing when I try to make a call like: SJphone - Asterisk - PBX (via mISDN) SJphone is talking ilbc to Asterisk, but obviously Asterisk talks alaw to the PBX, and Asterisk refuses to transcode. That seems odd to me, since I didn't do anything special to my install 1.2.10 and ilbc can be translated for me? Sorry I have no ideas, but wanted to let you know, that apparently it is just you. If you are running 1.2.10 you should update, as there was a bug that affected translations as I recall. Maybe that's the issue? Good Luck, Marty ___ --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] Re: can't transcode ilbc
Sorry! I think 1.2.12 had the bug I was referring to. Marty ___ --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] Queue AddQueueMember()
-Original Message- From: Douglas Garstang Sent: Friday, September 29, 2006 4:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Queue AddQueueMember() -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Thursday, September 28, 2006 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue AddQueueMember() On 9/28/06, Douglas Garstang [EMAIL PROTECTED] wrote: All, I've recently been told that the AgentCallBacklogin() application is buggy, and I should not use it. Apparently I should use AddQueueMember() instead. I see though that AddQueueMember() does not take the location to call back as an argument. We have remote agents that are available via PSTN access only. With AgentCallBackLogin() they can enter their PSTN phone number, and Asterisk will call them back at that number when they get a queue call. Can AddQueueMember() do that? Is AgentCallBackLogin() going to be deprecated at some point? Will AddQueueMember() be improved to match the call back functionality of AgentCallBackLogin()? AgentCallBackLogin() is deprecated beginning with 1.4. You can use AddQueueMember() in combination with the Local/ channel to do what you're looking to do above. The queue function AgentCallBackLogin() would take the name of an agent as an argument, and would log that agent into the queue they where associated with. We programmed our appearances on our Polycom phones to have a different appearance for each queue, and we'd send the caller id of the appearance as the name of the agent, thereby removing the need for the agent to enter their agent number. I see though that the AddQueueMember() function requires a queue name as an argument. That makes the dialplan logic more complex as we somehow have to magically send the queue name that this agent belongs to from the phone to the dialplan. I was really hoping for a response to this as I believe removing AgentCallbackLogin() is a -LOSS- of functionality! Doug. ___ --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] Re: WiFi SIP handset with Bluetooth required
On 2006-10-01 05:28:24 -0700, Andy Green [EMAIL PROTECTED] said: Hello, Can anyone point me in the right direction to source a WiFi SIP handset = that can also connect to a Bluetooth headset. I have a requirement for a hands free warehouse/distribution centre = setup using such devices and Asterisk I have checked the manufacturers websites that I know of but don't seem = to be able to find anything. I am not looking for a mobile phone network enabled device as there is = no requirement for it to be used away from the local WiFi network The Nokia e60 can do this, but you won't be using most of its function (ie gsm cell phone etc). The PocketPC suggestion should also work. You can also look at the HTC devices. Good Luck, Marty ___ --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] t1 voip to failover pri
On Mon, Oct 02, 2006 at 01:14:45PM -0400, Steve Glaus wrote: stan ford wrote: I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. Correcet. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. Yep. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to have a PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only? Well, yes and no. is anyone out there, using a VOIP only with no failover? We're using VOIP only, no failover. Furthermore we're using it on a cable internet connection. We have a cheap dsl connection for backups. It's been up for about 2 months now and has only been out twice for a small period of time. When that happens the DSL takes over. I don't pretend that this is in anyways comparable to PSTN service but it works pretty well for us. We have three locations. Two of which are set up the same way, the third just has 3 stations and just registers with one of the asterisk boxes at the other locations. Cool. I think when you're talking enterprise you definitely want to go with a t1 or two t1's for backup. (I don't really understand how a PRI gives you more reliability than a T circuit. They run over the same copper don't they??) They do. . For our purposes however (and I'd like to think I speak for a lot of mid size businesses with 50 employees) our setup works wonderful. It costs us about $600 all in all (internet access + VOIP) and that's a FAR cry from what we were paying through Covad before. Of course there always will be exceptions (People that need 100% guaranteed uptime), but for the size of our business this works. The only part that REALLY concerns me is our DID's. If our DID provider ever goes down we are screwed. Anyone know of any failsafes for THIS? I don't believe that you can port local DID's no. The easiest way to do it would be to leverage Local Number Portability, but this would require finding a LEC to serve you that a) could do that in realtime, and b) *would* do that in realtime. I'm not up on that state of the art, but I have people to ask. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] g729 Codec for AMD Sempron
Hi group,Can anyone help out in selecting the right codec to download from the digium site.Im using an AMD Sempron 2800+ CPU speed 1.6 GhzThanks in advanceDan ___ --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] Fax detection ...
Thanks Marco! I found NVFaxDetect before getting around to your post. It works a treat! Good call! no pun intended Marco Mouta [EMAIL PROTECTED] l.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 01/10/2006 23:09 Re: [asterisk-users] Fax detection ... Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Why don't you look for application NVfaxDetect ? are you using Digium boards? I've been using it sucessfully for fax reception! Look for it on voip wiki. On 10/1/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Oct 01, 2006 at 02:58:37PM -0700, Lee Howard wrote: Well, fax detection isn't entirely reliable anyway. Even if you assume that your fax detection feature and operation is flawless in properly detecting fax tones (and that most likely would be a specious assumption), not all calling fax machines send fax tones. So, y'know, that assertion gets made a lot. What's the turn rate of fax machines in the market? 3 years? 5? CNG tones are *well* over 10 years old, no? What percentage of fax calls are sent without CNG tones these days? Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think '87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Passing Arguments to FastAGI
How does one do this? Thanks, Doug. ___ --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] Fax detection ...
It all boils down to this: If they don't send a tone I won't get the fax. Its like my email now with DNS blacklists enabled. If they have a dial-up ADSL account they can't send me mail as my server denied them. Different technology, same problem. Whatever they invent next will, more than likely, have the same problem. If they really want me to receive whatever it is they want me to see ... they can post it :-) that nearly always works too! Phil. Lee Howard [EMAIL PROTECTED] van.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 02/10/2006 17:31 Re: [asterisk-users] Fax detection ... Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Jay R. Ashworth wrote: My personal experience is that I've never seen a consumer-grade fax machine with send-CNG turned off, and I don't *think* I've ever seen one on which there was a knob *to* turn it off; I would be less sure about fax modems -- those may have a knob, but I would expect it to default on. On fax modems the way to silent-dial (and I believe that this was a norm from early-on) to to add an @ at the end of the dialstring: ATDT5551212@. I would be very surprised to find any modern fax modem that does not have this capability. I don't know of any specific fax machine that has such a knob to turn CNG off. But my contention wasn't that it was consumer-grade fax machines that were the main culprit here, but rather fax servers (PCs with fax modems in them). And depending on what industry you are sampling, those may actually consitute a fair amount of the caller pool. (For example, some industry software - like insurance agent application software - will have built-in fax features that will use the PC's fax modem - and the application vendor may insist on that feature being used.) I cannot cite specific software that does it, but I suspect that most fax application developers are aware of the ability to silent-dial, and the reasons why it may be employed. As I've said before, one reason, as an example, is to make the modem capable of hearing ringback - so that it knows if the call has been answered or not (which itself is a unreliable endeavor). Another reason is to avoid annoying the receiver moreso on a call to a wrong-number. It's off the topic of silent dialing but on the topic of this thread... Brother fax machine manuals state that it is possible for them to erroniously detect certain voices or music as CNG tones, and if that becomes a problem to disable fax detection. And that's basically another point along the lines that say that fax detection is not completely reliable. Lee. ___ --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] t1 voip to failover pri
I wouldn't first presume that there is any law that states you have to run VoIP over a PRI. Other technologies exist such as SDSL - with some providers speaking of crazy prices such as £65pcm for2Mbs5:1 contention ration and £100 for 1:1 (U.K) - should be even less in the states if that's where you are. One option, but not the only one, would be to drop your pri when your contract ends and take up SDSL - and voila an initial saving, inyour case,of a 000 or more in the year.You could also have two SDSL lines for a little less than the price of the PRI. Both lineswould not only serve for High Availability -possibly even better availabilitythan single PRI- but could also, actively, both switch traffic, giving you 4Mbps of bandwidth for your VoIP, or if you choose, some other requirement while not required as failover- all for the price of less than one PRI.Then there is compression - 64k non negotiable, per channel forPRI, and flexible -i.e., less the 64k- for VoIP (International high quality Calls are transported at 16k),giving you the capacity to potentially service more traffic with less initial outlay.Other real cost efficiencies come in the form of the fact that IP-to-IP (local/national/international) calls are free. So if you have a lot of inter-branch communications, or communications you can switch on to IP,you can totally erradicate this cost - unlike with the PRI where you will still be subject to payment. Think like this - say I have two offices - one in london and the other New York. How much will I save by moving my calls on to VoIP with no per-time or call setup charges.Features related to OAMP,can also be faster and cheaper with you having a lot more power in your hands.In real senses, and with regards to reliability, you should take in to consideration the great moves currently being made by telecom companies (incumbents most especially), with regards to a complete shift to NGNs, which have a strong focus on ToIP. With new fiber (FTTP), new technology, etc, a lot of networks are highly reliable at the present moment - I guess this would also depend on where you are.The thing about it is that complete IP networks in terms of telecom now look inevitable. And whether you do it yourself or it is done for you - it is the way things, many expect, are going to be in the next 5 or so years. stan ford [EMAIL PROTECTED] wrote:I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to havea PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only?is anyone out there, using a VOIP only with no failover? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Now you can scan emails quickly with a reading pane. Get the new Yahoo! Mail.___ --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] t1 voip to failover pri
if i went with an SDSL line, don't those lines hook up to a common point, the DSLAM?i do like this idea of faling over not to a pri but another cheaper high speed line.adebayo omo-dare [EMAIL PROTECTED] wrote:I wouldn't first presume that there is any law that states you have to run VoIP over a PRI. Other technologies exist such as SDSL - with some providers speaking of crazy prices such as £65pcm for2Mbs5:1 contention ration and £100 for 1:1 (U.K) - should be even less in the states if that's where you are. One option, but not the only one, would be to drop your pri when your contract ends and take up SDSL - and voila an initial saving, inyour case,of a 000 or more in the year.You could also have two SDSL lines for a little less than the price of the PRI. Both lineswould not only serve for High Availability -possibly even better availabilitythan single PRI- but could also, actively, both switch traffic, giving you 4Mbps of bandwidth for your VoIP, or if you choose, some other requirement while not required as failover- all for the price of less than one PRI.Then there is compression - 64k non negotiable, per channel forPRI, and flexible -i.e., less the 64k- for VoIP (International high quality Calls are transported at 16k),giving you the capacity to potentially service more traffic with less initial outlay.Other real cost efficiencies come in the form of the fact that IP-to-IP (local/national/international) calls are free. So if you have a lot of inter-branch communications, or communications you can switch on to IP,you can totally erradicate this cost - unlike with the PRI where you will still be subject to payment. Think like this - say I have two offices - one in london and the other New York. How much will I save by moving my calls on to VoIP with no per-time or call setup charges.Features related to OAMP,can also be faster and cheaper with you having a lot more power in your hands.In real senses, and with regards to reliability, you should take in to consideration the great moves currently being made by telecom companies (incumbents most especially), with regards to a complete shift to NGNs, which have a strong focus on ToIP. With new fiber (FTTP), new technology, etc, a lot of networks are highly reliable at the present moment - I guess this would also depend on where you are.The thing about it is that complete IP networks in terms of telecom now look inevitable. And whether you do it yourself or it is done for you - it is the way things, many expect, are going to be in the next 5 or so years. stan ford [EMAIL PROTECTED] wrote:I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to havea PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only?is anyone out there, using a VOIP only with no failover? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Now you can scan emails quickly with a reading pane. Get the new Yahoo! Mail.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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] Fax detection ...
[EMAIL PROTECTED] wrote: It all boils down to this: If they don't send a tone I won't get the fax. And I certainly understand this approach. However, there are some situations where this is simply not suitable - where missing a fax costs money. Take, for example, the real estate industry where a fickle customer (isn't everyone fickle when dealing with real estate?) can be surveying mortgage quotes (frequently done by fax) and if you don't get their faxed authorization to pull their credit report and then you call them two days later to find out where it is ... well that customer may have already signed with another lender... and that just cost that agent 1-2% of the mortgage price (so that lost fax easily cost $2000 in lost opportunity). Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Minexpiry time - how to set this
I am trying to set a minimum expiry time. I have the latest trixbox installed and I have added minexpiry=60 in sip.conf. However my sniffer shows my client can still get any expiry that it wants without getting any rejection the server I tried to apply a patch I fond from digum but it is outdated and does not apply to 1.2 Any help would be great Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax detection ...
You can trick their machine into sending tones. The following code with
send tones that a terminating fax machine would normally respond with.
This will even force really old G2 fax machines to respond.
indications.conf:
faxrec = !2100/2600,!0/10,!1850/2600
[custom-fax-did]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,SetCallerID(${FROM_DID})
exten = s,4,Answer
exten = s,5,Playtones(faxrec)
exten = s,6,Wait(6)
exten = s,7,rxfax(${FAXFILE})
exten = s,8,system(tiff2pdf -p letter ${FAXFILE} -o ${FAXFILE}.pdf)
exten = s,9,system(mime-construct --to ${EMAILADDR} --subject Fax DID
${CALLERIDNUM} --attachment ${CALLERIDNUM}.pdf --type
application/pdf --file ${FAXFILE}.pdf)
exten = s,10,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten = s,11,Hangup
James Taylor
www.metrotel.net
- Original Message -
From: Lee Howard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 02, 2006 2:16 PM
Subject: Re: [asterisk-users] Fax detection ...
[EMAIL PROTECTED] wrote:
It all boils down to this: If they don't send a tone I won't get the fax.
And I certainly understand this approach. However, there are some
situations where this is simply not suitable - where missing a fax costs
money. Take, for example, the real estate industry where a fickle
customer (isn't everyone fickle when dealing with real estate?) can be
surveying mortgage quotes (frequently done by fax) and if you don't get
their faxed authorization to pull their credit report and then you call
them two days later to find out where it is ... well that customer may
have already signed with another lender... and that just cost that agent
1-2% of the mortgage price (so that lost fax easily cost $2000 in lost
opportunity).
Lee.
___
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Re: [asterisk-users] How do I reset a password?
Have you tried passwd-amp or passwd-maint from the command line? George On 9/30/06, Tom Vile [EMAIL PROTECTED] wrote: login as root and type help-aah and you will see a list of commands to change the admin password. On 9/30/06, Jim Lynch [EMAIL PROTECTED] wrote: I'm looking for the username/password to access the web gui for freepbx admin rather than the voicemail passwords. I need to reconfigure the extentions/ring groups. Thanks, Jim. Doug Lytle wrote: Jim Lynch wrote: I've forgotten the user/pw for my freepbx adnim. I'm using [EMAIL PROTECTED] Is there a way to discover them or reset them. I have root access to the system. I did a google search but that didn't help. On a normal installation the passwords are located in the voicemail.conf file located in /etc/asterisk/voicemail.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] extensions.conf strangeness
On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote: [invalid] exten = _X!,1,Answer() exten = _X!,2,Background(pbx-invalid) Are you sure that your invalid context is correctly written? I've never heard about this pattern match _X! As far as i know the wild card is the . So your invalid context should be: [invalid] exten = _X.,1,Answer() exten = _X.,2,Background(pbx-invalid) This may be the cause _X! means match the pattern as soon as it possibly could. If you use _X. then a timeout has to take place to see whether some other pattern might match. But your explanation still doesn't go into why it works differently in one context than another. I guess I'm going to have to assume that Asterisk dialplans are non-deterministic :-( Are there any debug tools which can show the thought process as a dial-plan is processed - for example, what patterns are tried and in what order? Thanks, Brian. ___ --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] Configuration / dialplan problem
I have my extensions.conf set up as follows:
exten = _Z.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _01.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _02.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0800.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0845.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0870.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _09.,1,Congestion()
exten = _00.,1,Congestion()
exten = _07.,1,Congestion()
(where nn are actually real digits).
I would expect this to let me dial the 07956nn numbers etc while
stopping dialing to other 07... numbers, but it seems to stop dialling
to any 07... number including the 3 specifically listed.
Any ideas?
Thanks
Mark
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[asterisk-users] Problems with Tormenta 2 quad card
Hello, I am trying to run Tormenta 2 Quad E1 (non-Digium clone) card on one of my asterisk box. I don't know why the card is not taking any interrupts: CPU0 CPU1 0:10996711067142IO-APIC-edge timer 1: 9 0IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 15: 1089986IO-APIC-edge ide1 169: 0 0 IO-APIC-level uhci_hcd 177: 0 0 IO-APIC-level uhci_hcd 185: 0 0 IO-APIC-level ehci_hcd 201: 0 0 IO-APIC-level Intel 82801DB-ICH4 209: 43132 0 IO-APIC-level eth0 217: 15916 10007 IO-APIC-level ioc0 225: 29 0 IO-APIC-level ioc1 233: 0 8 IO-APIC-level tor2 Card is configured properly, although any cable is connected to any port, a driver dosen't show any alarms. [EMAIL PROTECTED] zaptel-trunk]# cat /proc/zaptel/1 | head -2 Span 1: Tor2/0/1 Tormenta 2 (PCI) Quad E1 Card 0 Span 1 HDB3/CCS/CRC4 ClockSource [EMAIL PROTECTED] zaptel-trunk]# cat /proc/zaptel/2 | head -2 Span 2: Tor2/0/2 Tormenta 2 (PCI) Quad E1 Card 0 Span 2 HDB3/CCS/CRC4 [EMAIL PROTECTED] zaptel-trunk]# cat /proc/zaptel/3 | head -2 Span 3: Tor2/0/3 Tormenta 2 (PCI) Quad E1 Card 0 Span 3 HDB3/CCS/CRC4 [EMAIL PROTECTED] zaptel-trunk]# cat /proc/zaptel/4 | head -2 Span 4: Tor2/0/4 Tormenta 2 (PCI) Quad E1 Card 0 Span 4 HDB3/CCS/CRC4 I've compiled zaptel with WATCHDOG enabled and in dmesg I can see: Span Tor2/0/1 is dead with no revival Span Tor2/0/2 is dead with no revival Span Tor2/0/3 is dead with no revival Span Tor2/0/1 is alive! Span Tor2/0/2 is alive! Span Tor2/0/3 is alive! Span Tor2/0/1 is dead with no revival Span Tor2/0/2 is dead with no revival Span Tor2/0/3 is dead with no revival Span Tor2/0/4 is dead with no revival Does anyone can told me where is the problem ? Card is broken (I've two tor2 cards and both behaves the same way). -- Tomasz Paszkowski Systems engineer DCG Dominas Consulting Group Sp. z o.o. http://www.dcg.pl ___ --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] Fax detection ...
Interesting trick!
On the down side, won't sending this tone be pointless? If the receiver is
not sure a fax is calling, then he will BEEP every caller (even voice
calls). If the receiver is sure a fax is calling, why play the tones?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sent: Monday, October 02, 2006 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax detection ...
You can trick their machine into sending tones. The following code with
send tones that a terminating fax machine would normally respond with.
This will even force really old G2 fax machines to respond.
indications.conf:
faxrec = !2100/2600,!0/10,!1850/2600
[custom-fax-did]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,SetCallerID(${FROM_DID})
exten = s,4,Answer
exten = s,5,Playtones(faxrec)
exten = s,6,Wait(6)
exten = s,7,rxfax(${FAXFILE})
exten = s,8,system(tiff2pdf -p letter ${FAXFILE} -o ${FAXFILE}.pdf)
exten = s,9,system(mime-construct --to ${EMAILADDR} --subject Fax DID
${CALLERIDNUM} --attachment ${CALLERIDNUM}.pdf --type
application/pdf --file ${FAXFILE}.pdf)
exten = s,10,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten = s,11,Hangup
James Taylor
www.metrotel.net
- Original Message -
From: Lee Howard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 02, 2006 2:16 PM
Subject: Re: [asterisk-users] Fax detection ...
[EMAIL PROTECTED] wrote:
It all boils down to this: If they don't send a tone I won't get the fax.
And I certainly understand this approach. However, there are some
situations where this is simply not suitable - where missing a fax costs
money. Take, for example, the real estate industry where a fickle
customer (isn't everyone fickle when dealing with real estate?) can be
surveying mortgage quotes (frequently done by fax) and if you don't get
their faxed authorization to pull their credit report and then you call
them two days later to find out where it is ... well that customer may
have already signed with another lender... and that just cost that agent
1-2% of the mortgage price (so that lost fax easily cost $2000 in lost
opportunity).
Lee.
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[asterisk-users] Trunks and Outbound Routes
Hello - I have a small PBX setup for testing, and have put two small business accounts (from within my organization) on the PBX to see how things work out. I have two trunks two outbound routes setup and am using Teliax as my ITSP (two accounts; one for each account (different billing)). Outgoing Route1 is generic; lets anything out and is bound to TrunkA / ITSP Account 1. Outgoing Route2 requires a 9 to be dialed, and uses TrunkB, and is bound to ITSP Account 2. Is it possible that when I pick up extension 101, Asterisk detects that extension 101 belongs to TrunkB and uses TrunkB automatically for outbound calls, rather than having to dial 9 before the number? This functionality seems like a necessity when hosting multiple clients on one Asterisk box ... for obvious reasons. Thanks in advance,Dakota ___ --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 Quality / Echo / Problems
Try running the echo test from both the house side and the co (outside) side. That will let us know where the problem is. Post results. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Monday, October 02, 2006 6:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Quality / Echo / Problems Hi all I'm having a problem getting usable quality from my Asterisk setup. *SETUP* 2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones in the house and 2 x FXO to receive calls and in the future faxes. Gentoo Linux Here is what I've done so far (1) Moved theTDM 400p (FXS, , FXO, FXO) to it's own interrupt (It was sharing in the past) cat /proc/interrupts CPU0 0: 10236724 XT-PIC timer 1:486 XT-PIC i8042 2: 0 XT-PIC cascade 5: 40694267 XT-PIC wctdm== 10: 196233 XT-PIC eth0 12:225 XT-PIC i8042 14: 247177 XT-PIC ide0 15: 26 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 (2) I'm running the latest kernel uname -r 2.6.17- (3) I am running the latest Asterisk Asterisk-1.2.12.1 Libpri-1.2.3 zaptel-1.2.8 I compiled Zaptel with make clean ; make linux26 ; make install (4) ztmonitor has become my friend the Ring sends the VU meter off the chart, but the voice is below half way. I have tried changing the rxgain, txgain but that doesn't improve much. It raises the Volume and I can heard better. But the feedback (What I hear of myself in my analog headset) is off tone , too loud and poor (5) ztspeed reports Count: 254114 (Not sure if that is good or bad) ??? (6) zttest reports --- Results after 474 passes --- Best: 100.00 -- Worst: 92.578125 -- Average: 99.713476 (7) I have even run ./fxotune which generated /etc/fxotune.conf 3=3,0,0,0,0,0,0,0,0 4=4,0,0,0,0,0,0,0,0 Found http://www.voip-info.org/wiki/view/Asterisk+fxotune But the -d -b 3 doesn't work only -i and -s are allowed. PROBLEM The Call tone has a tin can sound (too much highs and not enough lows, for those with musical backgrounds) The Volume has improved. It did sound like I was talking behind my hand in front of my mouth, but not anymore. The is a static HISSS that randomly comes and goes and gets so loud that it drowns out whatever the calling party is trying to say. It can be heard on both ends but very Loud on the FXS connected phone Any Ideas What I can try next Thanks All Barry ___ --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] Fax detection ...
I've only used this on dedicated fax numbers.
I noticed that some fax machines didn't send tones and Asterisk didn't
detect.
They just sat there and looked at each other.
After playing the tones, the fax machines started sending and it worked.
James Taylor
www.metrotel.net
- Original Message -
From: Michelle Dupuis [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, October 02, 2006 5:04 PM
Subject: RE: [asterisk-users] Fax detection ...
Interesting trick!
On the down side, won't sending this tone be pointless? If the receiver
is
not sure a fax is calling, then he will BEEP every caller (even voice
calls). If the receiver is sure a fax is calling, why play the tones?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sent: Monday, October 02, 2006 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax detection ...
You can trick their machine into sending tones. The following code with
send tones that a terminating fax machine would normally respond with.
This will even force really old G2 fax machines to respond.
indications.conf:
faxrec = !2100/2600,!0/10,!1850/2600
[custom-fax-did]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,SetCallerID(${FROM_DID})
exten = s,4,Answer
exten = s,5,Playtones(faxrec)
exten = s,6,Wait(6)
exten = s,7,rxfax(${FAXFILE})
exten = s,8,system(tiff2pdf -p letter ${FAXFILE} -o ${FAXFILE}.pdf)
exten = s,9,system(mime-construct --to ${EMAILADDR} --subject Fax DID
${CALLERIDNUM} --attachment ${CALLERIDNUM}.pdf --type
application/pdf --file ${FAXFILE}.pdf)
exten = s,10,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten = s,11,Hangup
James Taylor
www.metrotel.net
- Original Message -
From: Lee Howard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 02, 2006 2:16 PM
Subject: Re: [asterisk-users] Fax detection ...
[EMAIL PROTECTED] wrote:
It all boils down to this: If they don't send a tone I won't get the
fax.
And I certainly understand this approach. However, there are some
situations where this is simply not suitable - where missing a fax costs
money. Take, for example, the real estate industry where a fickle
customer (isn't everyone fickle when dealing with real estate?) can be
surveying mortgage quotes (frequently done by fax) and if you don't get
their faxed authorization to pull their credit report and then you call
them two days later to find out where it is ... well that customer may
have already signed with another lender... and that just cost that agent
1-2% of the mortgage price (so that lost fax easily cost $2000 in lost
opportunity).
Lee.
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[asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?
I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1 firmware to get the new NAT keep-alive feature and the ability to watch more than a handful of buddy contacts but it appears to have broken the buddy-watch feature. Is anyone seeing this? Anybody know if it's a Polycom problem or something on the Asterisk end? I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather bone-headed move, I updated the firmware and Asterisk at the same time so I'm unable to tell which is the culprit. Curious, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --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] Re: WiFi SIP handset with Bluetooth required
If you don't need the Bluetooth bit then I can recommend someone to you. Drop me an email for more info -Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 03, 2006 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: WiFi SIP handset with Bluetooth required On 2006-10-01 05:28:24 -0700, Andy Green [EMAIL PROTECTED] said: Hello, Can anyone point me in the right direction to source a WiFi SIP handset = that can also connect to a Bluetooth headset. I have a requirement for a hands free warehouse/distribution centre = setup using such devices and Asterisk I have checked the manufacturers websites that I know of but don't seem = to be able to find anything. I am not looking for a mobile phone network enabled device as there is = no requirement for it to be used away from the local WiFi network The Nokia e60 can do this, but you won't be using most of its function (ie gsm cell phone etc). The PocketPC suggestion should also work. You can also look at the HTC devices. Good Luck, Marty ___ --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] Fax detection ...
Jay R. Ashworth wrote: On Mon, Oct 02, 2006 at 10:43:44AM +0800, Steve Underwood wrote: Jay R. Ashworth wrote: On Sun, Oct 01, 2006 at 02:58:37PM -0700, Lee Howard wrote: Well, fax detection isn't entirely reliable anyway. Even if you assume that your fax detection feature and operation is flawless in properly detecting fax tones (and that most likely would be a specious assumption), not all calling fax machines send fax tones. So, y'know, that assertion gets made a lot. What's the turn rate of fax machines in the market? 3 years? 5? CNG tones are *well* over 10 years old, no? What relevance does that have to CNG? It was a feature of the original spec 30 years ago. Well, perhaps I wasn't paying attention, but I thought that CNG tones *had as their purpose* making receive FAX detection trivial. That would tend to make the question on-point, would it not? It is the age of the machines which has no relevance. What percentage of fax calls are sent without CNG tones these days? Quite a lot. A large number of FAX machines have CNG turned off. On many machines, if select features like sharing a line between FAX and answering machine CNG, CED and various other useful behaviour might be disabled. My personal experience is that I've never seen a consumer-grade fax machine with send-CNG turned off, and I don't *think* I've ever seen one on which there was a knob *to* turn it off; I would be less sure about fax modems -- those may have a knob, but I would expect it to default on. Could you expand on what behaviour you think CNG breaks? Cause I'm not modeling it, mentally... I guess you touched many consumer grade fax machines, since *most* above the very basic ones can do this in some way. Some have really fun behaviour. I used to suffer some Olivetti ones that had several modes of calling and answering - Delay a while, to give someone a chance to pick up first; Pick up, but remain silent and see what the machine can hear; etc. Those Olivettis would only properly send a fax to another Olivetti when they were in straight forward standard mode. Now there's compatibility for you. :-) A lot of other machines offer similarly dumb modes of behaviour, but nothing quite so extreme as those. When you investigate one of these issues, and ask the user why the machine is not in simple answering mode, they are usually unaware it is not. These modes get set largely at random on installed fax machines. Now, it seems like these special modes should only affect answering. It would seem they are mostly about doing what Asterisk is doing - waiting silently for the 1100Hz tone. However, that's just too clean and simple for the fax industry. They do a bunch of other dumb stuff to make things more awkward, like call and only send something when they here 2100Hz. Steve Steve ___ --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] Spying a channel in a meetme
I'm using the ChanSpy command for monitor a conversation of a channel which is
in a meetme conference. All comunications go throught voip, with some voip
phones attached to the lan and an external voip providor in order to make
external calls.
The problem is that sometimes the spy call can hear the other persons of the
conference, but sometimes it works ok. Almost all conferences are only of two
channels.
exten=s,1,Chanspy(${SPYCHAN}|q)
I will try using monitor mode in meetme application, but I prefer Chanspy
because I can spy the call always, not only when it is in a conference.
You can join the Meetme room muted...
--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [asterisk-users] Fax detection ...
On Tue, Oct 03, 2006 at 08:44:16AM +0800, Steve Underwood wrote: So, y'know, that assertion gets made a lot. What's the turn rate of fax machines in the market? 3 years? 5? CNG tones are *well* over 10 years old, no? What relevance does that have to CNG? It was a feature of the original spec 30 years ago. Well, perhaps I wasn't paying attention, but I thought that CNG tones *had as their purpose* making receive FAX detection trivial. That would tend to make the question on-point, would it not? It is the age of the machines which has no relevance. Ah. My early memories of CNG tones suggested that early fax machines did not actually send them. My personal experience is that I've never seen a consumer-grade fax machine with send-CNG turned off, and I don't *think* I've ever seen one on which there was a knob *to* turn it off; I would be less sure about fax modems -- those may have a knob, but I would expect it to default on. Could you expand on what behaviour you think CNG breaks? Cause I'm not modeling it, mentally... I guess you touched many consumer grade fax machines, since *most* above the very basic ones can do this in some way. Yeah, the only real pro grade fax machines I've ever run across was a Panafax I have that uses 3-inch core paper rolls, and I never really got that working. Some have really fun behaviour. I used to suffer some Olivetti ones that had several modes of calling and answering - Delay a while, to give someone a chance to pick up first; Pick up, but remain silent and see what the machine can hear; etc. Those Olivettis would only properly send a fax to another Olivetti when they were in straight forward standard mode. Now there's compatibility for you. :-) Yeah: I'm so compatible with everyone else that I'm not compatible with myself is kinda dumb. A lot of other machines offer similarly dumb modes of behaviour, but nothing quite so extreme as those. When you investigate one of these issues, and ask the user why the machine is not in simple answering mode, they are usually unaware it is not. These modes get set largely at random on installed fax machines. Yay. Now, it seems like these special modes should only affect answering. It would seem they are mostly about doing what Asterisk is doing - waiting silently for the 1100Hz tone. However, that's just too clean and simple for the fax industry. They do a bunch of other dumb stuff to make things more awkward, like call and only send something when they here 2100Hz. What fun. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] Polycom Buddy Watch Broken with 2.0.1 Firmware?
Yes, we saw the same problem. We opened a ticket with our reseller, who escalated it to Polycom. Here's what the reseller said... We escalated this up to Polycom. They said that they had seen a problem with the display not updating with Asterisk, and they are going to see if your issue is a match for a known issue or if this is a new different issue. I expect an update from the later today (Monday) or tomorrow. I'll let you know as soon as I hear from them. Guess that snuck through Polycom's QA process. You can watch more than a handful of buddies with version 1.6.7. :) Doug. -Original Message- From: Paul Dugas [mailto:[EMAIL PROTECTED] Sent: Mon 10/2/2006 6:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware? I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1 firmware to get the new NAT keep-alive feature and the ability to watch more than a handful of buddy contacts but it appears to have broken the buddy-watch feature. Is anyone seeing this? Anybody know if it's a Polycom problem or something on the Asterisk end? I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather bone-headed move, I updated the firmware and Asterisk at the same time so I'm unable to tell which is the culprit. Curious, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --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] Polycom Buddy Watch Broken with 2.0.1 Firmware?
I did the same thing with the Polycom's - upgraded all mine from 1.6.x to 2.0.1 but I had great success and no problem with the buddy watch / presence feature --- if anything, it works a little better. Whats your mac-address-directory.xml configuration file look like? Did you make any changes to the mac-address-phone.cfg file? do you have the line of: up.useDirectoryNames=1 feature.1.name=presence feature.1.enabled=1 In the config? Scott Higginbotham Systems / Network Operations Manager 215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Dugas Sent: Monday, October 02, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware? I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1 firmware to get the new NAT keep-alive feature and the ability to watch more than a handful of buddy contacts but it appears to have broken the buddy-watch feature. Is anyone seeing this? Anybody know if it's a Polycom problem or something on the Asterisk end? I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather bone-headed move, I updated the firmware and Asterisk at the same time so I'm unable to tell which is the culprit. Curious, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --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] Polycom Buddy Watch Broken with 2.0.1 Firmware?
I had some weird flaky-ness after upgrading to the latest. Did a format file system and let it reload from scratch. Works like a charm. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Higginbotham Sent: Monday, October 02, 2006 10:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware? I did the same thing with the Polycom's - upgraded all mine from 1.6.x to 2.0.1 but I had great success and no problem with the buddy watch / presence feature --- if anything, it works a little better. Whats your mac-address-directory.xml configuration file look like? Did you make any changes to the mac-address-phone.cfg file? do you have the line of: up.useDirectoryNames=1 feature.1.name=presence feature.1.enabled=1 In the config? Scott Higginbotham Systems / Network Operations Manager 215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Dugas Sent: Monday, October 02, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware? I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1 firmware to get the new NAT keep-alive feature and the ability to watch more than a handful of buddy contacts but it appears to have broken the buddy-watch feature. Is anyone seeing this? Anybody know if it's a Polycom problem or something on the Asterisk end? I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather bone-headed move, I updated the firmware and Asterisk at the same time so I'm unable to tell which is the culprit. Curious, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --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] Polycom Buddy Watch Broken with 2.0.1 Firmware?
Install went fine. No troubles other than this and it'd be minor if one of the reasons for the update wasn't to expand the number of buddies allowed on the IP601+sidecards we're adding for the attendant. Ugh... Anyway, directory entries haven't changed: ?xml version=1.0 standalone=yes?^M !-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05 $ --directory item_list item lnDoe/ln fnJane/fn ct1001/ct sd1/sd bw1/bw /item /item_list /directory The config entries you referred to are set in my global sip.cfg and apply to all of the units. Looks right to me. Did some sniffing and Asterisk is sending a NOTIFY like so: ... ?xml version=1.0 encoding=ISO-8859-1? presence xmlns=urn:ietf:params:xml:ns:pidf xmlns:pp=urn:ietf:params:xml:ns:pidf:person xmlns:es=urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status xmlns:ep=urn:ietf:params:xml:ns:pidf:rpid:rpid-person entity=sip:[EMAIL PROTECTED] pp:personstatus /status/pp:person noteReady/note tuple id=1001 contact priority=1sip:[EMAIL PROTECTED]/contact statusbasicopen/basic/status /tuple /presence --- Extension Changed 1001 new state Idle for Notify User x1002 pbx*CLI Hmmm On Mon, 2006-10-02 at 22:14 -0400, Scott Higginbotham wrote: I did the same thing with the Polycom's - upgraded all mine from 1.6.x to 2.0.1 but I had great success and no problem with the buddy watch / presence feature --- if anything, it works a little better. Whats your mac-address-directory.xml configuration file look like? Did you make any changes to the mac-address-phone.cfg file? do you have the line of: up.useDirectoryNames=1 feature.1.name=presence feature.1.enabled=1 In the config? Scott Higginbotham Systems / Network Operations Manager 215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Dugas Sent: Monday, October 02, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware? I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1 firmware to get the new NAT keep-alive feature and the ability to watch more than a handful of buddy contacts but it appears to have broken the buddy-watch feature. Is anyone seeing this? Anybody know if it's a Polycom problem or something on the Asterisk end? I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather bone-headed move, I updated the firmware and Asterisk at the same time so I'm unable to tell which is the culprit. Curious, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --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 -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --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] TDM2400P wiring.
I just received my first TDM2400 card I tried searching and couldn't find anything on this. I have 2 FXO modules with this card, it came with one modlule in the slot marked as slot 6, so I put the other in slot 5. Since I don't have an Amphenol connector/cable and a 66 block at the moment I can't realy test it. I'm therefore turning here for help. Which slot on the TDM24xxP card is Pair 1 thru 4 on the 66 block? Thank you ___ --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] tools/techniques/metrics for measurement of end-point quality
I am looking for advise on tools/techniques/metrics that are commonly used to measure quality of device/end-points. Any pointers will very helpful. 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] Configuration / dialplan problem
There are a few things to look at.
First off, you have a lot of wildcard testing that is probably throwing
the dial plan off. For example, you have the following:
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07.,1,Congestion()
If I left it in this order what would happen? From what I understand it
is nautral to think in that order, but really Asterisk is going to sort
the extensions something like this:
exten = _07.,1,Congestion()
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
So now say you dial 07545865143254/8564, it will go to the Congestion
application every time.
What I would do is comment out the wildcard searches and see if that
resolves the problem. If so, try putting all the wildcard tests in an
include and see if that helps.
Take a look at these to articles as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
Also just out of observation, why all the testing? Seems to me you could
streamline that code down a bit more. For example, the 01 and 02 tests.
If you know they are dialing N number of digits, make the test
_01XX, so you know they have to dial a certain amount of digits
to be a valid call. Why send a 4 digit number out your trunk if you know
it isn't going anywhere? If you need to dial '0' then 10 digits, try this:
_01NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
_02NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
_07956X,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) 3
etc.
Hopefully that will help,
Kevin
Mark Muffett wrote:
I have my extensions.conf set up as follows:
exten = _Z.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _01.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _02.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0800.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0845.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0870.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _09.,1,Congestion()
exten = _00.,1,Congestion()
exten = _07.,1,Congestion()
(where nn are actually real digits).
I would expect this to let me dial the 07956nn numbers etc while
stopping dialing to other 07... numbers, but it seems to stop dialling
to any 07... number including the 3 specifically listed.
Any ideas?
Thanks
Mark
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Re: [asterisk-users] G726 prompts
On 10/3/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: It seems unreasonably difficult to get a list of the supported formats, but does sox (http://sox.sourceforge.net/) do what you need? Cheers, -- jra hey Jay, thanks but I am not sure what to tell sox as my output format to be. I must admit, I missed it the first time I was thinking about using it. Should've looked at the man page. This time I looked at it again It seems like I could convert ulaw pcm files into the adpcm format. But what output format do I choose? I tried raw but that doesn't work. I tried using the following like sox -r 8000 -c 1 input.ul output.raw anyone know what the correct parameters are for using sox to convert pcm or ulaw prompts into g726? thanks \R ___ --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
