[asterisk-users] where can I found documentation about channel drivers
Hi! I am looking for authoritative documentation about channel driver options, e.g. 'n' and 'j' option for chan_local or the SIP channel option to set a specific To: header. Is there such documentation available (except on the mailing list and the voip-info wiki (which is usually very old))? thanks Klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support using different ip for rtp?
I think this is not possible. If you take a look at main/rtp.c there is no config option for an IP address. regards klaus Jun Yin schrieb: some vendors(like alcatel-lucent) developed a kind of sip proxy which includes two parts: one sip signaling module and one or more voice modules. voice modules are responsible for receiving/sending voice traffic(RTP). each voice module has its own IP. so , when the sip signaling part sends out invite packet, it has sip ip in its sip content and different RTP ip in SDP content. (also for 200OK) Now I'm trying to do a test to simulate that product with asterisk. I hope asterisk can sends out different rtp address based on user or domain name. Based on network side, there are many ways to do it: we can configure the network card with multiple IPs, one for SIP and others for RTP. or , we can setup multiple network cards for the asterisk server, one card is for sip signaling and other cards for rtp traffic connecting to different carriers. I think this diagram is reasonable but I was surprised that asterisk does not support it. Maybe asterisk can do this by special configuration? or, there is other free sip proxy software can do this? Thanks. Message: 10 Date: Wed, 25 Jun 2008 05:15:29 -0400 From: Raj Jain [EMAIL PROTECTED] Subject: Re: [asterisk-users] Can asterisk support using different ip for rtp? 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 On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin [EMAIL PROTECTED] wrote: Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? SIP is decoupled from RTP, so they can emanate from different IP addresses. Can you present a scenario where this will make sense (in the context where Asterisk is anchoring the media) ? -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where can I found documentation about channel drivers
Klaus Darilion schrieb: Hi! I am looking for authoritative documentation about channel driver options, e.g. 'n' and 'j' option for chan_local or the SIP channel option to set a specific To: header. Answer myself: I have found the documentation about chan_local's options in doc/tex/asterisk.pdf. But no information about SIP options :-( Is there such documentation available (except on the mailing list and the voip-info wiki (which is usually very old))? thanks Klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where can I found documentation about channel drivers
26 jun 2008 kl. 10.17 skrev Klaus Darilion: Hi! I am looking for authoritative documentation about channel driver options, e.g. 'n' and 'j' option for chan_local or the SIP channel option to set a specific To: header. Is there such documentation available (except on the mailing list and the voip-info wiki (which is usually very old))? There was an attempt to document all the dialstrings - which are the ones you look for - by John Todd a long time ago. I haven't seen any progress. For SIP, I think it's quite well documented in sip.conf.sample: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=co /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue with different music for each caller
Hi, I tried this before I ask here on the list. In 1.2 SetMusicOnHold did not work. The Moh class defined in queues.conf is overwriting any SetMusicOnHold values of the caller channel. You can see this if you use periodic announce, the Moh call is printed in the CLI and is allways the class defines in queues.conf. I have now the choice to switch to 1.4 or implement for every music an single queue. best regards Thomas On Wednesday 25 June 2008 06:55, Martin Schrott - thinking:systems wrote: Hello Thomas, no problem. In asterisk 1.6 use SetMusicOnHold(musiconholdname) then it will work in older Asterisk versions! br, Martin - Original Message - From: Thomas Winter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 24, 2008 5:50 PM Subject: Re: [asterisk-users] Queue with different music for each caller On Tuesday 24 June 2008 15:22, Martin Schrott - thinking:systems wrote: Hello Thomas you can use different music for each caller if you like. in extensions.conf you can set the music class. exten = s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller) Hi Martin, thanks for your suggestion, I forgot to notice that Iam still using 1.2.X Jun 24 17:45:31 ERROR[17784]: pbx.c:1437 ast_func_write: Function CHANNEL not registered So, this didnt work for me. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound video Calls
Hi all,
I am trying to make an outbound video call to a mobile from asterisk.
however it keeps failing.
I can make inbound calls from a mobile and view video.
I am using x-lite to initiate the outbound call, however I have tried using
the management interface as well (action: etc...) and result is the same.
normal voice outbound calls work fine.
Circuit is a q931 30 channel from telewest (virgin media).
Any pointers would be appreciated.
below is pri debug output and relevant conf entries.
// BEGIN //
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/paul-081ff260,
video_test_out|666|1) in new stack
-- Goto (video_test_out,666,1)
-- Executing [EMAIL PROTECTED]:1] Set(SIP/paul-081ff260,
CHANNEL(transfercapability)=VIDEO) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/paul-081ff260,
CHANNEL(userinformationlayer1)=38) in new stack
-- Executing [EMAIL PROTECTED]:3] h324m_gw(SIP/paul-081ff260,
[EMAIL PROTECTED]) in new stack
[Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't know
any of 0x2000 formats
-- Executing [EMAIL PROTECTED]:1]
h324m_call(Local/[EMAIL PROTECTED],2,
[EMAIL PROTECTED]) in new stack
-- Executing [EMAIL PROTECTED]:1]
Set(Local/[EMAIL PROTECTED],2,
CHANNEL(transfercapability)=VIDEO) in new stack
-- Executing [EMAIL PROTECTED]:2]
NoOp(Local/[EMAIL PROTECTED],2, transfer=VIDEO) in
new stack
-- Executing [EMAIL PROTECTED]:3]
Set(Local/[EMAIL PROTECTED],2,
CHANNEL(userinformationlayer1)=38) in new stack
-- Executing [EMAIL PROTECTED]:4]
NoOp(Local/[EMAIL PROTECTED],2, ul1=38) in new stack
-- Executing [EMAIL PROTECTED]:5]
Dial(Local/[EMAIL PROTECTED],2,
Zap/g0/07525029025|40|tTkK) in new stack
-- Making new call for cr 32771
-- digital call, setting user information layer 1 to 38 (0x26)
-- Requested transfer capability: 0x18 - VIDEO
Protocol Discriminator: Q.931 (8) len=38
Call Ref: len= 2 (reference 3/0x3) (Originator)
Message type: SETUP (5)
[04 03 88 90 a6]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (8)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: H.223 and
H.245 (38)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 1 ]
[6c 06 41 80 70 61 75 6c]
Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) 'paul' ]
[70 0c c1 30 37 35 32 35 30 32 39 30 32 35]
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ]
[a1]CLI
Sending Complete (len= 1)
q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call
Initiated)
-- Called g0/07525029025
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 3/0x3) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 03 80 e4 04]
Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: User (0)
Ext: 1 Cause: Invalid information element contents
(100), class = Protocol Error (e.g. unknown message) (6) ]
Cause data 1: 04 (4)
-- Processing IE 8 (cs0, Cause)
q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null)
-- Channel 0/1, span 1 got hangup, cause 100
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:6]
Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
== Spawn extension (video_test_out_context, dialcell, 6) exited non-zero
on 'Local/[EMAIL PROTECTED],2'
== Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2'
status is 'UNKNOWN'
== Spawn extension (video_test_out, 666, 3) exited non-zero on
'SIP/paul-081ff260'
// END //
extensions.conf:
[video_test_out]
exten = 666,1,Set(CHANNEL(transfercapability)=VIDEO)
exten = 666,n,Set(CHANNEL(userinformationlayer1)=38)
exten = 666,n,h324m_gw([EMAIL PROTECTED])
exten = 666,n,Hangup
[video_test_out_context]
exten = s,1,h324m_call([EMAIL PROTECTED])
exten = dialcell,1,Set(CHANNEL(transfercapability)=VIDEO)
exten = dialcell,n,NoOp(transfer=${CHANNEL(transfercapability)})
exten = dialcell,n,Set(CHANNEL(userinformationlayer1)=38)
exten = dialcell,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
exten = dialcell,n,Dial(Zap/g0/07x,40,tTkK)
exten = dialcell,n,Hangup()
exten = t,1,Goto(s,2)
sip.conf:
[general]
context=sip_in
allowoverlap=no
Re: [asterisk-users] GotoIfTime Function
Finally did it but only one more problem, I want it to ring once before going to the context or playing the background message. [day_menu] exten = s,1,Answer() exten = s,2,Background(welcome-message) exten = s,3,Dial(SIP/5960,200,rt) ; week day goes to Philadelphia Office [weekend__menu] exten = s,1,Answer() exten = s,2,Background(welcome-message) exten = s,3,Dial(SIP/5961,200,rt) ; weekend goes to Delaware Office [night_menu] exten = s,1,Answer() exten = s,2,Background(officeclosed) exten = s,3,Hangup ; ;incoming exten = 1866x,1,GotoIfTime(8:00-18:00|mon-sun|*|*?day_menu,s,1) exten = 1866x,n,Goto(night_menu,s,1) On Tue, Jun 24, 2008 at 6:35 AM, broadband Voice [EMAIL PROTECTED] wrote: I googled some information on voip.org. Its my fault though and implemented the sample implementation without creating the context an the include statements. On Mon, Jun 23, 2008 at 10:33 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: If any docs were the cause of this (very important) misconception, maybe the docs could be reworded. Do you remember what caused you to think that context was created automatically? broadband Voice wrote: fc7234153*CLI dialplan show open There is no existence of 'open' context I was under the impression that this was part of the Asterisk default libraries. I will create the context then and also add the include files. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where can I found documentation about channel drivers
Johansson Olle E schrieb: 26 jun 2008 kl. 10.17 skrev Klaus Darilion: Hi! I am looking for authoritative documentation about channel driver options, e.g. 'n' and 'j' option for chan_local or the SIP channel option to set a specific To: header. Is there such documentation available (except on the mailing list and the voip-info wiki (which is usually very old))? There was an attempt to document all the dialstrings - which are the ones you look for - by John Todd a long time ago. I haven't seen any progress. For SIP, I think it's quite well documented in sip.conf.sample: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=co Thanks, this one I was looking for. ; All of these dial strings specify the SIP request URI. ; In addition, you can specify a specific To: header by adding an ; exclamation mark after the dial string, like ; ; SIP/[EMAIL PROTECTED]@edvina.net IMO, a sip.conf is not the probper place for this, but better than nothing. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound video Calls
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too. Maybe the switch wants to have it in Bearer Capability and LCC (I once had such a switch). Another reason could be that the telco blocks video calls. regards klaus PS: use the asterisk-video mailing lists Asterisk Users schrieb: Hi all, I am trying to make an outbound video call to a mobile from asterisk. however it keeps failing. I can make inbound calls from a mobile and view video. I am using x-lite to initiate the outbound call, however I have tried using the management interface as well (action: etc...) and result is the same. normal voice outbound calls work fine. Circuit is a q931 30 channel from telewest (virgin media). Any pointers would be appreciated. below is pri debug output and relevant conf entries. // BEGIN // -- Executing [EMAIL PROTECTED]:1] Goto(SIP/paul-081ff260, video_test_out|666|1) in new stack -- Goto (video_test_out,666,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/paul-081ff260, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/paul-081ff260, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:3] h324m_gw(SIP/paul-081ff260, [EMAIL PROTECTED]) in new stack [Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't know any of 0x2000 formats -- Executing [EMAIL PROTECTED]:1] h324m_call(Local/[EMAIL PROTECTED],2, [EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Local/[EMAIL PROTECTED],2, transfer=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:3] Set(Local/[EMAIL PROTECTED],2, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Local/[EMAIL PROTECTED],2, ul1=38) in new stack -- Executing [EMAIL PROTECTED]:5] Dial(Local/[EMAIL PROTECTED],2, Zap/g0/07525029025|40|tTkK) in new stack -- Making new call for cr 32771 -- digital call, setting user information layer 1 to 38 (0x26) -- Requested transfer capability: 0x18 - VIDEO Protocol Discriminator: Q.931 (8) len=38 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 88 90 a6] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: H.223 and H.245 (38) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 06 41 80 70 61 75 6c] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'paul' ] [70 0c c1 30 37 35 32 35 30 32 39 30 32 35] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ] [a1]CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call Initiated) -- Called g0/07525029025 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 80 e4 04] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 04 (4) -- Processing IE 8 (cs0, Cause) q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null) -- Channel 0/1, span 1 got hangup, cause 100 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:6] Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (video_test_out_context, dialcell, 6) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2' status is 'UNKNOWN' == Spawn extension (video_test_out, 666, 3) exited non-zero on 'SIP/paul-081ff260' // END // extensions.conf: [video_test_out] exten = 666,1,Set(CHANNEL(transfercapability)=VIDEO) exten = 666,n,Set(CHANNEL(userinformationlayer1)=38) exten =
Re: [asterisk-users] Chef-secretary scenario
Grygoriy Dobrovolskyy schrieb:
You have 2 choices to pickup someone's phone with snom's
1: imagine yourself prefix of pickup, let's say 4
exten=4XX,1,Pickup([EMAIL PROTECTED])
so if u call 4 + phone number you will pickup that one.
Second you can add pickupgroup=number for each phone you want to be in
the group, and add a dtmf button on snom with string set in
features.conf (pickup)
To answer you next question: Yes i would be nice to pickup a phone by
pressing blinking button on snom,and use that button fo call when ext is
out of use, but i dont know the way to make asterisk doing that.
It works (at least with asterisk 1.2) by patching Asterisk with
bristuff. bristuff implement this feature.
regards
klaus
2008/6/25 Vazquez David [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Though I wonder...
The scenario is as follows:
I have 4 phones with the following extensions:
11 (SIP/11)
12 (SIP/12)
13 (SIP/13)
15 (SIP/15)
Whenever SIP/11 receives a call, it hints the other phones. Is it
possible to pick up that call from one of them?
The relevant part of my extensions.conf looks as this now:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s,hint,SIP/11
exten = 11,hint,SIP/12SIP/13SIP/15
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return
to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/
busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
Thanks :-)
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[asterisk-users] Fw: Outbound video Calls
Hi, You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too. Maybe the switch wants to have it in Bearer Capability and LCC (I once had such a switch). Just applied the patch, failed again. can you tell me if theres anything more i need to add to the conf file to signal in LLC as well ? Another reason could be that the telco blocks video calls. They keep telling me that there shouldnt be a problem, however they are not the brightest bunch :-) regards klaus PS: use the asterisk-video mailing lists Just have :-) Asterisk Users schrieb: Hi all, I am trying to make an outbound video call to a mobile from asterisk. however it keeps failing. I can make inbound calls from a mobile and view video. I am using x-lite to initiate the outbound call, however I have tried using the management interface as well (action: etc...) and result is the same. normal voice outbound calls work fine. Circuit is a q931 30 channel from telewest (virgin media). Any pointers would be appreciated. below is pri debug output and relevant conf entries. // BEGIN // -- Executing [EMAIL PROTECTED]:1] Goto(SIP/paul-081ff260, video_test_out|666|1) in new stack -- Goto (video_test_out,666,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/paul-081ff260, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/paul-081ff260, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:3] h324m_gw(SIP/paul-081ff260, [EMAIL PROTECTED]) in new stack [Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't know any of 0x2000 formats -- Executing [EMAIL PROTECTED]:1] h324m_call(Local/[EMAIL PROTECTED],2, [EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Local/[EMAIL PROTECTED],2, transfer=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:3] Set(Local/[EMAIL PROTECTED],2, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Local/[EMAIL PROTECTED],2, ul1=38) in new stack -- Executing [EMAIL PROTECTED]:5] Dial(Local/[EMAIL PROTECTED],2, Zap/g0/07525029025|40|tTkK) in new stack -- Making new call for cr 32771 -- digital call, setting user information layer 1 to 38 (0x26) -- Requested transfer capability: 0x18 - VIDEO Protocol Discriminator: Q.931 (8) len=38 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 88 90 a6] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: H.223 and H.245 (38) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 06 41 80 70 61 75 6c] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'paul' ] [70 0c c1 30 37 35 32 35 30 32 39 30 32 35] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ] [a1]CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call Initiated) -- Called g0/07525029025 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 80 e4 04] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 04 (4) -- Processing IE 8 (cs0, Cause) q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null) -- Channel 0/1, span 1 got hangup, cause 100 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:6] Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (video_test_out_context, dialcell, 6) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2' status is 'UNKNOWN' == Spawn extension (video_test_out, 666, 3) exited
Re: [asterisk-users] Number portability in other parts of the world.
Steve Kennedy a écrit : [...] Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? In the UK numbers are portable, though the telco wanting the number must have a porting agreement with the telco that has the number. Not all telcos have porting agreements. Same in France. If the number is an original France Telecom one, no problem. If the number was _already_ ported, can be a problem. In all other cases, I would suggest you to check if there is an agreement between telco. In Poland, not possible today, should be end 2008/begining 2009. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
Well, I think I've solved the problem, just to let you know, I've just added the Answer() app before the Call(Zap/N) app. Thanks a lot to Yannick Lam Hang of Sangoma Technologies for suggesting that!!! On Wed, Jun 25, 2008 at 9:04 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Well, I have new information if anyone can/want to help me... (Please read all the previous messages in this email) If I call a number that can't hear me at all (calling from inside my network using a Grandstream GXP-2000 phone through Asterisk) and then I put this call on hold for a second and then I take again the call, then the callee start hearing me, :s Any ideas??? Thanks in advance... -- Nacho Linux Counter #156439 On Tue, Jun 17, 2008 at 7:50 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: I've been playing around in order to find something new and I've found this: I have created an IVR for test purposes, then I've placed a call from my sip phone using one of my telco lines to another of my telco lines attached to the PBX, in this situation I'm using two FXO channels, one for the outgoing call and another for the incoming call. Then I have created an extension in this IVR in order to make an echo test and I've used MixMonitor() to record the audio of the test. When I dial this extension I never can hear my echoed voice, but when I listen to the recording the audio have a lot of artifacts and the busy and dial tone are almost inaudible, the same effect that happens when you play to almost identical audio files, so I can presume that it is the same audio wave but out of phase (meaning the echo is working, I think). I don't know if this can be happening because of the Hardware Echo Canceler on my Remora A400D. If I call the extension of the echo test directly from my SIP phone without using any telco line (SIP -- IP -- Asterisk) then the test works just fine. Another test I've made is, during a call with the one way audio problem, I have used the ZapBarge() application to hear what's happening on the Zap Channel (from another SIP phone on my network). In this case I heard the callee complaining that he/she can't hear anything and I can't hear the caller (which is on the same network of my phone). In this case the caller can hear the callee. I have grabbed the sip debug messages of this call from the asterisk CLI and is attached (compressed) to this email. Well, thanks again for any comment/response... -- Nacho Linux Counter #156439 On Tue, Jun 17, 2008 at 5:14 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi Steve and the rest of the list, On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T My Asterisk Server has two NIC with a channel bonding setup (Balance TLB) connected to the same switch, and it does not have any firewall rule. I'm attaching a file with the output of sip set debug on the CLI of a call in this situation. Although calls made with SIP phones have this strange behavior, when I place a call with an analog phone connected to a FXS port of the same TDM card (see below for full description) this does not happen. Thanks, any help will be really appreciated... -- Nacho Linux Counter #156439 On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi list, I'm having trouble with calls placed to the PSTN (through a TDM card), sometimes (a lot indeed) when I dial a number the callee party can't hear me at all. My setup is: Asterisk 1.4.20.1 Zaptel 1.4.11 libpri 1.4.4 Wanpipe 3.2.4 I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel 2.4.16.60-0.23-smp I'm using the ulaw audio codec. There is no NAT between the Asterisk Server and the Phones (the phone and the server are in the same network segment). What can it be??? Thanks in advance for any help/comment... -- Raul Linux Counter #156439 Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T -- Nacho Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs. SKINNY
On Thu, 2008-06-26 at 06:15 -0500, [EMAIL PROTECTED] wrote: Date: Wed, 25 Jun 2008 23:41:18 +0200 From: Michiel van Baak [EMAIL PROTECTED] Subject: Re: [asterisk-users] SIP vs. SKINNY To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On 14:16, Wed 25 Jun 08, Joe Carroll wrote: Can anyone comment on the performance benefits when comparing sip to skinny ? Most cisco phones work better with the skinny firmware. That is not true when connecting to asterisk though. It all depends on the version of asterisk you are running. I have a setup with over 20 skinny phones on asterisk -trunk and that works great. Specially after today, now that chan_skinny supports transfers. If you are running 1.4 I'm not sure what is best. It basically depends on what you are doing with the phones. In my home setup it worked great, but in my business I have to run trunk for the phones to be as workable as the sip variant. The skinny firmware has some neat stuff like XML push etc. Dont know how the current SIP firmware is doing, as I have not run it in over 2 years now. YMMV Does Skinny let Cisco 79xx phones act as extensions *across the Internet* to a remote Asterisk server? Does SIP? How do the different SCCP channels compare to the chan_skinny support, in Asterisk 1.6? Is there a better guide than http://www.voip-info.org/wiki/view/chan_skinny to getting chan_skinny working best with Asterisk and Cisco 79xx phones? Michiel van Baak -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disconnection from caller did not recognized
Dear, I am using ser + asterisk, for outgoing calls, my problem is that the session was not closed if the caller says bye. can u help me ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_zap.c:4747 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1
Anybody else get theses warning? [Jun 26 10:08:55] WARNING[3158]: chan_zap.c:4747 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk With Web meetme
Hi I followed this howto http://www.voip-info.org/wiki/view/MeetMe-Web-Control and http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html to install web meetme with asterisk, I know the meetme module is included however I need to be able to ban and mute users as well. All of the installation went fine however when I do call a conference number I create using the interface all I get is service unavailable, I did run asterisk in verbose mode that did not make me any smarter. I added to extensions.conf the following [confserv] ;Make sure you change 1199 to your conference bridge extension(s) ;more information on this can be found at the asterisk web site. exten = 121212,1,Answer exten = 121212,n,Wait(3) exten = 121212,n,CBMysql() exten = 121212,n,Hangup Where 121212 is an existing extension, I really dont get it this all of the documentation available but I surely missed something here..any hints please ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: Outbound video Calls
you also need (as stated in the bug report) the patch 10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch from http://bugs.digium.com/view.php?id=10217 This enables LCC in chan_zap. Is this was done some time ago I do not remember anymore who it is activated, I think you have to add the h324m=lcc option to zapata.conf I remember one scenario where H324M signaling was required to be in Bearer Capabilite AND Low Layer Compatibility. I think you can easily extend the patches to signal both versions at the same time. Always take a look at the outgoing SETUP message to see if it contains LCC. PS: Please dump an incoming SETUP message for a video call - does it contain LCC too? regards klaus Asterisk Users schrieb: Hi, You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too. Maybe the switch wants to have it in Bearer Capability and LCC (I once had such a switch). Just applied the patch, failed again. can you tell me if theres anything more i need to add to the conf file to signal in LLC as well ? Another reason could be that the telco blocks video calls. They keep telling me that there shouldnt be a problem, however they are not the brightest bunch :-) regards klaus PS: use the asterisk-video mailing lists Just have :-) Asterisk Users schrieb: Hi all, I am trying to make an outbound video call to a mobile from asterisk. however it keeps failing. I can make inbound calls from a mobile and view video. I am using x-lite to initiate the outbound call, however I have tried using the management interface as well (action: etc...) and result is the same. normal voice outbound calls work fine. Circuit is a q931 30 channel from telewest (virgin media). Any pointers would be appreciated. below is pri debug output and relevant conf entries. // BEGIN // -- Executing [EMAIL PROTECTED]:1] Goto(SIP/paul-081ff260, video_test_out|666|1) in new stack -- Goto (video_test_out,666,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/paul-081ff260, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/paul-081ff260, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:3] h324m_gw(SIP/paul-081ff260, [EMAIL PROTECTED]) in new stack [Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't know any of 0x2000 formats -- Executing [EMAIL PROTECTED]:1] h324m_call(Local/[EMAIL PROTECTED],2, [EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Local/[EMAIL PROTECTED],2, transfer=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:3] Set(Local/[EMAIL PROTECTED],2, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Local/[EMAIL PROTECTED],2, ul1=38) in new stack -- Executing [EMAIL PROTECTED]:5] Dial(Local/[EMAIL PROTECTED],2, Zap/g0/07525029025|40|tTkK) in new stack -- Making new call for cr 32771 -- digital call, setting user information layer 1 to 38 (0x26) -- Requested transfer capability: 0x18 - VIDEO Protocol Discriminator: Q.931 (8) len=38 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 88 90 a6] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: H.223 and H.245 (38) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 06 41 80 70 61 75 6c] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'paul' ] [70 0c c1 30 37 35 32 35 30 32 39 30 32 35] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ] [a1]CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call Initiated) -- Called g0/07525029025 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 80 e4 04] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 04 (4) --
[asterisk-users] Error while Compiling zaptel-1.4.11
Hello All, This is my third freshly installed and updated CentOS 5.1 with installed Digium 4-port Analog card and while compiling Zaptel I am getting this error. If I run ./install_preq test and ./install_preq install it says Install Successfully. Error = CC [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/wcte12xp.o CC [M] /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /usr/src/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /usr/src/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here make[4]: *** [/usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.11/kernel/xpp] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.11/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.6.el5-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.11' make: *** [all] Error 2 [EMAIL PROTECTED] zaptel-1.4.11]# Can anyone help... Cheers, Nitesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_zap.c:4747 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1
Yes, it's happening to me too [Jun 26 07:54:56] WARNING[24659] chan_zap.c: CallerID returned with error on channel 'Zap/3-1' [Jun 26 07:54:57] WARNING[24659] chan_zap.c: Ring/Off-hook in strange state 6 on channel 3 Mostly of the time this two messages comes together. The other situation in which this message appears is on a PSTN line that is dead (no tone but some signaling is coming because if you call it Asterisk detects the ring but can't pickup the line) I have no problems so far because of this, so if you get some strange behavior and suspect on this please post and I will check for it too. -- Raul Linux Counter #156439 On Fri, Jun 27, 2008 at 9:40 AM, Paul Belanger [EMAIL PROTECTED] wrote: Anybody else get theses warning? [Jun 26 10:08:55] WARNING[3158]: chan_zap.c:4747 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk With Web meetme
Ali wrote: I followed this howto http://www.voip-info.org/wiki/view/MeetMe-Web-Control and http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html to install web meetme with asterisk, I know the meetme module is included however I need to be able to ban and mute users as well. All of the installation went fine however when I do call a conference number I create using the interface all I get is service unavailable, I did run asterisk in verbose mode that did not make me any smarter. I added to extensions.conf the following [confserv] ;Make sure you change 1199 to your conference bridge extension(s) ;more information on this can be found at the asterisk web site. exten = 121212,1,Answer exten = 121212,n,Wait(3) exten = 121212,n,CBMysql() exten = 121212,n,Hangup Where 121212 is an existing extension, I really dont get it this all of the documentation available but I surely missed something here..any hints please ? Let's start with the easy stuff, if confserv included in the context that the phone has access to? What is the output of the command CLIcb mysql status? What version of Asterisk and Web-MeetMe are you using? Do you have a timing source (ztdummy or PSTN interface card)? Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Users Conference June 27th @ 12 Noon EDT Scaling and Clustering
This Friday June 27th at Noon EDT, JR Richardson is joining us to talk
about scaling asterisk by clustering and server specialization. JR has
authored multiple documents on the subject but I'm unclear as to
whether he intended these to be published, so I'll wait to hear about
that.
Many participants have asked to have someone guest on these issues so
please be there and bring your questions.
See http://VoipUsersConference.org
Dial in this Friday to find out more.
IRC.Freenode.net #voip-users-conference
PSTN;: Call (724) 444-7444 and enter 22622# 1#
Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#)) ; use #1 if you
do not join Talkshoe
DNS: ts.x2z.eu resolves to the above IP
http://food4wine.ning.com has news, forums, blogs, etc
RSS http://feeds.feedburner.com/AstUser
Trademarks are copyright their various owners.
___
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
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Re: [asterisk-users] VoIP Users Conference June 27th @ 12 Noon EDT Scaling and Clustering
Hello, If am connecting a digium E1 card to a PSTN Switch in the same equipment room would I need an echo canceller? wouldnt the Switch handle echo cancellation for dial-in users? Responses would be appreciated. BR, Robor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo Cancelation
Hello, If am connecting a digium E1 card to a PSTN Switch in the same equipment room would I need an echo canceller? wouldnt the Switch handle echo cancellation for dial-in users? Responses would be appreciated. BR, Robor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancelation
On Thu, Jun 26, 2008 at 12:17 PM, Robor Oghene [EMAIL PROTECTED] wrote: Hello, If am connecting a digium E1 card to a PSTN Switch in the same equipment room would I need an echo canceller? wouldnt the Switch handle echo cancellation for dial-in users? Responses would be appreciated. BR, Robor Switch is very generic and you give no real details. But, I would say you should be fine based on the tiny bit of info provided. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number portability in other parts of the world.
I think it would be a good idea to start an item in the Wiki about this. Can anyone else chime in for their countries?? Others in the EU, Eastern, Far East? So Far I have: Australia: PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and PSTN to Cell are NOT OK.Dean Collins Poland: Not Today but possibly in 2009 Daniel UK: Portable if Telco has a porting agreement. Not all Telco have agreements in place. Steve Kennedy France: Porting from France Telcom to another provider not an issue, however if porting between other Telco's, Telco's must have porting agreement between them.Randulo -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Administrator TOOTAI Sent: Thursday, June 26, 2008 8:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Number portability in other parts of the world. Steve Kennedy a écrit : [...] Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? In the UK numbers are portable, though the telco wanting the number must have a porting agreement with the telco that has the number. Not all telcos have porting agreements. Same in France. If the number is an original France Telecom one, no problem. If the number was _already_ ported, can be a problem. In all other cases, I would suggest you to check if there is an agreement between telco. In Poland, not possible today, should be end 2008/begining 2009. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
Same here. Some of our clients upgraded from 1.4.18.1 to 1.4.21. After some time CLI stops responding and no calls are possible. Killall -9 is the only way to solve (get out) of this situation till next time it hangs. Example CLI screenshot: http://193.138.191.205/packets/asterisk1.4.21_noresponse.jpg Back to 1.4.18.1 (1.4.19.x is even more broken: http://lists.digium.com/pipermail/asterisk-users/2008-April/209342.html). Regards, Mindaugas Kezys http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Wednesday, June 25, 2008 7:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Major problem with 1.4.21 asterisk Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number portability in other parts of the world.
On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote: I think it would be a good idea to start an item in the Wiki about this. Can anyone else chime in for their countries?? Others in the EU, Eastern, Far East? So Far I have: Australia:PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and PSTN to Cell are NOT OK.Dean Collins Poland: Not Today but possibly in 2009 Daniel UK: Portable if Telco has a porting agreement. Not all Telco have agreements in place. Steve Kennedy France: Porting from France Telcom to another provider not an issue, however if porting between other Telco's, Telco's must have porting agreement between them. Randulo In the UK numbers can be ported between fixed operators and mobile operators, but not (yet) between mobile and fixed (but then the distinction is blurring). Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup channel
Hi all, I am getting a weird error here. When i send a call to a sip peer on one of our servers i get a 'Nobody picked up in -1 ms' immediately following the SIP INVITE then the call hangs up. I do not have a timeout in the Dial, if i send the call to a different peer the call works fine. I am running 1.2 SVN 2006-02-22 Here is the dial statement used: Executing Dial(SIP/1ST LEG, SIP/2ND CALL LEG||t) in new stack __ Not happy with your email address?. Get the one you really want - millions of new email addresses available now at Yahoo! http://uk.docs.yahoo.com/ymail/new.html___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Console/dsp in 1.4.X
When using console/dsp is that play only? Is it play/record mode? If so how can I make it play only? When I play wave files on a machine with aplay everything is fine. (no record) When I use asterisk and console/dsp I am getting seg faults in alsa-lib. I want to make sure there is NO record action coming back just play. Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On 6/26/08, Grey Man [EMAIL PROTECTED] wrote: On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote: This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! Hi, I just wanted to say that we are working on testing our current functionality. We don't use attended transfers, but would like at some point. So, I'll try to report within next week if something else is broken. Hi murf, From some preliminary testing on the CDRfix4 branch it looks like the CDRs for attended transfers are now correct which is fantastic. For blind transfers the CDR for the first call leg is still incorrect with the duration only being recorded up until the point the transfer occurs. What's wrong with that? This fits perfectly for my needs. Is there a way how to exploit this? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon: Early Bird Special ends next week
Astricon 2008 is less than three months away - the Early Bird discounts will expire on the last day of the month, which is next Tuesday - please get your registrations in by then to get up to $100 off the normal rates. Making hotel reservations now is also a good idea, since while there is a good supply of rooms at the conference hotel, the supply is limited at the conference rate ($134/night.) The 2008 Astricon is really shaping up to be the most tech-heavy conference ever! We've received a great list of speakers, and there are talks that really seem to be getting into the details of Asterisk's new features, how to implement large-scale services, and coverage of some great third-party applications in the Asterisk ecosystem. What makes Astricon great is the speakers, but also the participants. The ability to talk in person with people who are working in the same areas that you are, who have solved the same problems you're encountering, or who you've met online but never face-to-face - these are some of the most valuable parts of the conference. The informal parts of the conference are where you make connections, figure out code or implementation problems, or solve business issues that otherwise would be difficult or impossible to handle outside of such a concentrated group of similarly-minded people. Here is a tiny random sample of the 50+ topics we've got in the agenda: - A Carrier Grade VoIP Project with Asterisk. - OpenR2 in Asterisk - MFC/R2 free of headaches or your money back - Clustering Methods with Asterisk - Druid: Open Source Unified Communications - Asterisk Checks Into The Hotel - ISDN PRI Capabilities and the Asterisk Implementation - ATT SIP Trunk Compatibility Testing for Asterisk - Selling Asterisk-based Phone Systems In The Legacy World - CEL: an introduction to Asterisk's new call logging mechanism - Intro to Unified Communications: Two words, many challenges. - Carrier Class Routing using Asterisk ExternalIVR and the Griffin Routing Engine - Asterisk, meet Lua: An Introduction to pbx_lua - Measuring Signal Quality in Hybrid Systems (VOIP/PSTN) Hope to see you in September! - The Digium Astricon Staff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote: On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote: This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! Hi murf, From some preliminary testing on the CDRfix4 branch it looks like the CDRs for attended transfers are now correct which is fantastic. For blind transfers the CDR for the first call leg is still incorrect with the duration only being recorded up until the point the transfer occurs. I did a blind xfer with my snom360, and got these two cdrs with **TRUNK**: Eventlist: 1. 101 dahdi (used to be zap) phone picked up and 200 is dialed for the snom360 2. 200 (snom360) picks up and answers the call 3. 200 (snom360) hits the Transfer button (101 gets MOH), dials 202 4. 200 (snom360) hits the checkmark button to send off the call (101 starts hearing ringing, 200 starts getting congestion). 5. 202 (eyebeam) answers (101 202 are connected) 6. 101 or 202 hang up. Conversation finished. fxs.01 101,101,200,extension,DAHDI/1-1,SIP/snom360-082c3f68,Dial,SIP/snom360,30,2008-06-26 11:04:08,2008-06-26 11:04:12,2008-06-26 11:05:56,108,104,ANSWERED,DOCUMENTATION,,1214499848.11,, fxs.01 101,101,201,extension,DAHDI/1-1,SIP/murf-eyebeam-082d95d8,Dial,SIP/polycom430SIP/murf-eyebeam,30,2008-06-26 11:06:06,2008-06-26 11:06:12,2008-06-26 11:06:56,50,44,ANSWERED,DOCUMENTATION,,1214499966.13,, Here are the two CDR's with their recorded event times: CDR start answer end 112 3 245 6 above, I called into the snom360, and hit the Transfer button, dialed 201, and got congestion (101 gets moh until I hit the check key), and hung up the snom (200). 201, the eyebeam, rings, I answer. 101 and 201 are connected. 101 hangs up, and the conversation ended. THE SAME PROCEDURE ON THE CDRfix6 branch: fxs.01 101,101,200,extension,DAHDI/1-1,SIP/snom360-0829e2d0,Dial,SIP/snom360,30,Tt,2008-06-26 12:16:37,2008-06-26 12:16:44,2008-06-26 12:17:01,24,17,ANSWERED,DOCUMENTATION,,1214504197.4,, fxs.01 101,101,202,extension,DAHDI/1-1,SIP/murf-eyebeam-082c2b70,Dial,SIP/murf-eyebeam,30,Tt,2008-06-26 12:17:01,2008-06-26 12:17:14,2008-06-26 12:17:49,48,35,ANSWERED,DOCUMENTATION,,1214504197.4,, CDR start answer end 112 4 245 6 Well, time 3 does get lost, but I thought it might be nice to be able to link 1 2 by the coincident times and say, hey, that looks like a blind transfer! One point of dissatisfaction I have with these is the fact that SIP/snom dialed the second CDR, not DAHDI/1. But, if I change it, you won't know that DAHDI/1 was the guy that murf-eyebeam was talking to... tough choices. So, I take it from your above words, that you'd like the 1,2,3; 4,5,6; times on the two CDR's? Can anyone lab this up for 1.2; I don't have enough phones, and I'm not eager to reconfigure the ones I've got for just one test ! For people on the list following this bug my company got stung by this in the last week so there now appear to be some people out there actively looking for Asterisk systems to exploit. The incident for us was a user using attended transfers to place free calls through a 1.2 system. In the past we have had normal users stumble across the problem but in this case it was a directed attempt. So if like us you are a provider and use Asterisk and are required to support transfers it would be highly advisable to keep a close eye on things! Won't it be pleasant to slip in the fix and watch these guys get billed for calls they were thinking would be free! murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Nice, i will search for that.
2008/6/26 Klaus Darilion [EMAIL PROTECTED]:
Grygoriy Dobrovolskyy schrieb:
You have 2 choices to pickup someone's phone with snom's
1: imagine yourself prefix of pickup, let's say 4
exten=4XX,1,Pickup([EMAIL PROTECTED])
so if u call 4 + phone number you will pickup that one.
Second you can add pickupgroup=number for each phone you want to be in
the group, and add a dtmf button on snom with string set in
features.conf (pickup)
To answer you next question: Yes i would be nice to pickup a phone by
pressing blinking button on snom,and use that button fo call when ext is
out of use, but i dont know the way to make asterisk doing that.
It works (at least with asterisk 1.2) by patching Asterisk with
bristuff. bristuff implement this feature.
regards
klaus
2008/6/25 Vazquez David [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Though I wonder...
The scenario is as follows:
I have 4 phones with the following extensions:
11 (SIP/11)
12 (SIP/12)
13 (SIP/13)
15 (SIP/15)
Whenever SIP/11 receives a call, it hints the other phones. Is it
possible to pick up that call from one of them?
The relevant part of my extensions.conf looks as this now:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as
well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s,hint,SIP/11
exten = 11,hint,SIP/12SIP/13SIP/15
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return
to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail
w/
busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to
start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
Thanks :-)
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Re: [asterisk-users] Major problem with 1.4.21 asterisk
If I remember correctly there was a security patch released after 1.4.19, I think that's shwy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 26, 2008 12:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk Just out of curiosity, why did you feel they needed an upgrade? Thanks, Steve On Thu, Jun 26, 2008 at 12:01 AM, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hopefully the other guy with the problem can test it because this is a production server and the client is already upset about the problems this caused for a day or two till I realized what the issue is so I cant risk it. Maybe I can off hours if he cant though. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, June 25, 2008 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. Please try the patch in bug number 12903: http://bugs.digium.com/view.php?id=12903 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number portability in other parts of the world.
Well, here in Venezuela there is no way to port out numbers between Telcos (as far as i know) On Fri, Jun 27, 2008 at 12:28 PM, Steve Kennedy [EMAIL PROTECTED] wrote: On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote: I think it would be a good idea to start an item in the Wiki about this. Can anyone else chime in for their countries?? Others in the EU, Eastern, Far East? So Far I have: Australia:PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and PSTN to Cell are NOT OK.Dean Collins Poland: Not Today but possibly in 2009 Daniel UK: Portable if Telco has a porting agreement. Not all Telco have agreements in place. Steve Kennedy France: Porting from France Telcom to another provider not an issue, however if porting between other Telco's, Telco's must have porting agreement between them. Randulo In the UK numbers can be ported between fixed operators and mobile operators, but not (yet) between mobile and fixed (but then the distinction is blurring). Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nacho Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queues and MEMBERINTERFACE for AGI script
Hi, iam using and queue and starting an AGI script after caller connected to agent. How to find out in the script the connected agent, MEMBERINTERFACE seemed to be not work, either as variable in the queue command and also not in the AGI script. How to found out which agent is connected to calling channel? I try to avoid to using LOCAL channels, because I like the function ringinuse. regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On Thu, Jun 26, 2008 at 8:21 PM, Steve Murphy [EMAIL PROTECTED] wrote: On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote: Hi murf, CDR start answer end 112 4 245 6 Well, time 3 does get lost, but I thought it might be nice to be able to link 1 2 by the coincident times and say, hey, that looks like a blind transfer! One point of dissatisfaction I have with these is the fact that SIP/snom dialed the second CDR, not DAHDI/1. But, if I change it, you won't know that DAHDI/1 was the guy that murf-eyebeam was talking to... tough choices. So, I take it from your above words, that you'd like the 1,2,3; 4,5,6; times on the two CDR's? If i've understood your call flow correctly the CDR's required are 1,2,6 and 4,5,6. The key point being that the first call made is up until both call legs are hungup (which is 6) whereas the CDR is reporting its duration as the time up until the blind transfer was initiated (which is 3). As far as using the CDRs to identify that a blind transfer has taken place my opinion would be that that is a secondary concern compared to getting the call records accurate. There seem to be a lot of cases where people are experiencing pain because of the incorrect CDRs for their billing but I'm yet to see a post where someone is kicking up a fuss because they can't easily identify whether a particular CDR was involved in a transfer. It's would be a nice to have whereas incorrect durations on the CDRs cost money. Can anyone lab this up for 1.2; I don't have enough phones, and I'm not eager to reconfigure the ones I've got for just one test ! Do you mean compare the differences between the CRDfix4 branch and 1.2? At the moment the blind transfer CDRs are the same for 1.2, 1.4 and CDRfix4 with all being incorrect in the same spot which is the duration on the first call leg. In case it's of any help if you have a Windows box available I have a tool that can initiate SIP calls and carry out blind and attended transfers with Asterisk. It does make testing a lot easier, I got tired of playing hopscoth on my phones as well, now I just click a button. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
Yes I do remember now, I believe that there was a security vunerability in 1.4.19 and below that was addressed, that is why I updated. Do you ask because you want to know if you should upgrade yours or to give me one of those you shouldn't upgrade a production server if its not needed and working fine. I ask because if it's the former, I would be glad to answer any other questions you have regarding upgrading. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Thursday, June 26, 2008 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk If I remember correctly there was a security patch released after 1.4.19, I think that's shwy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 26, 2008 12:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk Just out of curiosity, why did you feel they needed an upgrade? Thanks, Steve On Thu, Jun 26, 2008 at 12:01 AM, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hopefully the other guy with the problem can test it because this is a production server and the client is already upset about the problems this caused for a day or two till I realized what the issue is so I cant risk it. Maybe I can off hours if he cant though. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, June 25, 2008 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. Please try the patch in bug number 12903: http://bugs.digium.com/view.php?id=12903 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] SIP/IAX2 Provider with fallback dialing?
Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, /sf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
We're personally located in a small office based in Manhattan. Would need DIDs for the greater Manhattan area. But it sounds like Speakup is the type of service we're looking for that would cater to us domestically. On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
I use Vitelity strictly for fall back to my cell (and testing). Thanks, Steve T On Thu, Jun 26, 2008 at 5:56 PM, Steve Finkelstein [EMAIL PROTECTED] wrote: We're personally located in a small office based in Manhattan. Would need DIDs for the greater Manhattan area. But it sounds like Speakup is the type of service we're looking for that would cater to us domestically. On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
I think Voicepulse is out of NYC... not sure if they have failover though... but they have iax2 and sip. http://connect.voicepulse.com/ is their asterisk page. Fred Posner Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com FWD#: 902963 On Jun 26, 2008, at 5:56 PM, Steve Finkelstein wrote: We're personally located in a small office based in Manhattan. Would need DIDs for the greater Manhattan area. But it sounds like Speakup is the type of service we're looking for that would cater to us domestically. On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
VoicePulse looks awesome, but they do not have the feature I need ... which is to be able to dial my mobile phone in the event my asterisk box or the Internet goes kablunk. On Thu, Jun 26, 2008 at 6:07 PM, Fred Posner [EMAIL PROTECTED] wrote: I think Voicepulse is out of NYC... not sure if they have failover though... but they have iax2 and sip. http://connect.voicepulse.com/ is their asterisk page. Fred Posner Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com FWD#: 902963 On Jun 26, 2008, at 5:56 PM, Steve Finkelstein wrote: We're personally located in a small office based in Manhattan. Would need DIDs for the greater Manhattan area. But it sounds like Speakup is the type of service we're looking for that would cater to us domestically. On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] start valgrind and asterisk via init.d script
List, Anybody have a script around that will do this? We have to run valgrind and asterisk to help troubleshoot a bug in the tracker. Since we do not know how to reproduce the error, we'd like to run them from an init.d script (simalar to safe_asterisk), email on crash, and restart asterisk. Ideas? Thanks, PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral ... Swift... weird result
Asterisk 1.2, and Cepstral 5, Allison voice. I execute: swift Please enter your pin. -o please-enter-your-pin.ulaw -p audio/channels=1,audio/encoding=ulaw,audio/sampling-rate=8000 then copy it up to /var/lib/asterisk/sounds, and Play() the file. The sound file seems corrupted. All I hear is 'please' or 'please' followed by the rest of the sentence said so fast I almost can't hear it. I've tried other various of the -p option to swift, same results. Also tried generating a wav file and converting to ulaw with sox, same result. I did this once before and it worked. What am I doing wrong? Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is it possible? 1 VOIP Provider Multiple registrations to multiple inbound contexts
The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been told that the PEER can be made to match DID or account name but I'm not sure how to do this. In other words how to match a registration to a peer or inbound context other that the single defined default. I've also been told back in the asterisk 1.2 days that it was not possible. I just recently upgraded to the latest 1.4 and am wondering if there have been updates to make this work. In the past I always had to bring all calls into a default context and then use GOTO to get the inbound call where it needed to go. Do I still need to do this today? or is there a better way to make this actually work and come into the proper context. I'm confused on how to get multiple registrations to be associated with any more than one default registration inbound context, or somehow to associate them with the peer of choice. I've tried amongst other things appending /[EMAIL PROTECTED] approach to a register: line which does not work. I've also tried adding register=yes inside a peer which seems to do nothing meaning it does not cause the system to register with the VOIP provider or even try to register. Thanks for your help! -Steve Here's a small example of what I am working with: Without getting into too much detail and showing Please realize I have everything else working well except getting an inbound call into a context that I choose. So I've not provided any more detail showing that those actually exist in the dialplan.. those work and I have working s and test extensions in my dialplan contexts that work as well :-) register = 734111:[EMAIL PROTECTED] ; standard inbound to 's' in default inbound context ; would like to be able to direct that inbound call ; into context1 instead of common default defined register = 734111:[EMAIL PROTECTED]/[EMAIL PROTECTED] ;example of me trying to get inbound call to 102 in context: context2 [peer-1] type=peer context=context1 secret=testpassword username=734111 fromuser=734111 fromdomain=sip.exampleprovider.net host=sip.exampleprovider.net ;register=yes usereqphone=yes insecure=very nat=yes canreinvite=yes ;call-limit=5 [peer-2] type=peer context=context2 secret=testpassword username=734111 fromuser=734111 fromdomain=sip.exampleprovider.net host=sip.exampleprovider.net ;register=yes usereqphone=yes insecure=very nat=yes canreinvite=yes ;call-limit=5 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues and MEMBERINTERFACE for AGI script
Thomas Winter wrote:
Hi,
iam using and queue and starting an AGI script after caller connected to
agent.
How to find out in the script the connected agent, MEMBERINTERFACE seemed to
be not work, either as variable in the queue command and also not in the AGI
script.
How to found out which agent is connected to calling channel?
I try to avoid to using LOCAL channels, because I like the function ringinuse.
regards
Thomas
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Thomas,
In queues.conf you need to set the variable setinterfacevar=yes
You'd then pass the AGI to the Queue application with something like this:
exten = some_extension,n,Queue(somequeue,some.agi)
Then within the AGI you'd retrieve the variable like this:
my $memberinterface = $AGI-get_variable('MEMBERINTERFACE'); # Perl example.
Hope this helps.
-Dave
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[asterisk-users] Asterisk, POTS and plain handsets
Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook, asterisk reads it as an incoming call and starts handling it accordingly, running through the context for that channel as if an incoming call was detected. I'd like asterisk to act like just another handset on the line or an answering machine and not do anything if a handset is used on that same line. It seems like maybe it's a voltage issue where asterisk or the zaptel module is sensing a voltage change on the line and so is doing what it thinks it should do. I'd like to know how to dumb it down or make it less sensitive to the changes, if indeed that is the cause. I've tried various combinations of asterisk versions and zap module versions and various combinations of phone lines to the card and, well, everything else that I can think of. So I'm hoping that someone out there has used an asterisk set up like this and maybe encountered the same things that I'm seeing. I'd be happy to post any configs that someone might find relevant. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, POTS and plain handsets
On Thu, Jun 26, 2008 at 9:11 PM, Steve [EMAIL PROTECTED] wrote: Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook, asterisk reads it as an incoming call and starts handling it accordingly, running through the context for that channel as if an incoming call was detected. I'd like asterisk to act like just another handset on the line or an answering machine and not do anything if a handset is used on that same line. It seems like maybe it's a voltage issue where asterisk or the zaptel module is sensing a voltage change on the line and so is doing what it thinks it should do. I'd like to know how to dumb it down or make it less sensitive to the changes, if indeed that is the cause. I've tried various combinations of asterisk versions and zap module versions and various combinations of phone lines to the card and, well, everything else that I can think of. So I'm hoping that someone out there has used an asterisk set up like this and maybe encountered the same things that I'm seeing. I'd be happy to post any configs that someone might find relevant. Steve Post the output from Asterisk's CLI. I think maybe your contexts are overlapping or are the same. It should say something to the effect of Starting simple switch Check what context your FXO channels are in, something like context=from-verizon and then check the context of your FXS (plain telephones), they should be in a different context such as context=to-phones. Also, make sure you have immediate=no Then check your dialplan and make sure those contexts do what you want and you are not accidentally including a context where it should not be. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DNS Query Overload
I'm finding that my Asterisk server is bombarding my DNS servers with lookups like the following: Queries 5060-b7bfce38: type A, class IN Name: 5060-b7bfce38 Type: A (Host address) Class: IN (0x0001) One call alone has a handful of requests to our server, simply looking for an A record for something like '5060-b7bfce38' (listed above). The DNS server immediately responds with No such name. I use only SIP on my box, and even if I just have the call go to hangup it does this. My SIP.conf contains 'srvlookup=no' in the general section. Any thoughts or suggestions? Best regards, Mik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is it possible? 1 VOIP Provider Multiple registrations to multiple inbound contexts
On Fri, Jun 27, 2008 at 2:20 AM, Steve Gladden [EMAIL PROTECTED] wrote: In other words how to match a registration to a peer or inbound context other that the single defined default. I've also been told back in the asterisk 1.2 days that it was not possible. Not true. When you register the /1234 on the end of the line sends it to that extension in the context you specified in the peer entry with context=. You don't mention what exactly happens when a call comes in on one of the DID? Unless the service provider has a non-standard method of calling your asterisk, it should work as you expected. Also, you could read the sample file that has every possible line in peer entries commented and explains all the possible terms. hth, r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
