[asterisk-users] Dial international number over dahdi trunk
Hi all,
I am trying to perform the following outgoing call:
exten = _49.,1,Log(NOTICE,Dialing German number: ${EXTEN})
same = n,Set(route=DAHDI/g1/00${EXTEN})
same = n,Dial(${route})
exten = _0049.,1,Goto(${EXTEN:2},1)
exten = _01149.,1,Goto(${EXTEN:3},1)
exten = _+49.,1,Goto(${EXTEN:1},1)
But this is not working. I have also tried changing the route to:
same = n,Set(route=DAHDI/g1/${EXTEN})
With similar results:
server1*CLI
...
-- Goto (ctx-carriers,4917,1)
-- Executing [4917@ctx-carriers:1]
Log(SIP/sipserver3-006c, NOTICE,Dialing German number:
4917) in new stack
[Jul 18 07:18:53] NOTICE[19497]: Ext. 4917:1 @ ctx-carriers:
Dialing German number: 4917
-- Executing [4917@ctx-carriers:2]
Set(SIP/sipserver3-006c, route=DAHDI/g1/004917) in new stack
-- Executing [4917@ctx-carriers:3]
Dial(SIP/sipserver3-006c, DAHDI/g1/004917) in new stack
-- Called DAHDI/g1/004917
-- Hungup 'DAHDI/63-1'
== Everyone is busy/congested at this time (1:0/0/1)
(the number dialed is not really 4917, but a real, working mobile
german number)
How can I dial international numbers via DAHDI? (in case it matters, my SS7
trunk provider is Telefonica España)
Thanks!
Daniel Gonzalez
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[asterisk-users] How to get 2 CDR Records of 2 outgoing calls bridge
Hi all,
I need 2 CDR Records of below 2 numbers for outgoing calls, detail is given
as below:
*9688*
*= Call file : outbound call generate through below file*
Test.call
==
Channel: local/s@outgoing/n
WaitTime: 45
Context: outgoing_ivrs
Extension: s
Priority: 1
Set: contact_no=96
extensions.conf
[outgoing]
exten = s,1,NoOP(- First LEG CALL from call file for outgoing:
Channel-${CHANNEL} -)
*same = n,Dial(DAHDI/g0/${contact_no},45)*
same = n,Noop(* DIALSTATUS-${DIALSTATUS}*)
same = n,hangup
exten = h,1,NoOP(Call hangup === outgoing context)
same = n,NoOP(calldate2=${CDR(calldate)},src2=${CDR(src)},dst2=${CDR(dst)})
same =
n,NoOP(channel2=${CDR(channel)},dstchannel2=${CDR(dstchannel)}start2=${CDR(start)})
same =
n,NoOP(end2=${CDR(end)},duration2=${CDR(duration)},billsec=${CDR(billsec)},disposition2=${CDR(disposition)})
[outgoing_ivrs]
exten = s,1,Noop(- Second LEG CALL from call file for outgoing_ivrs:
Channel-${CHANNEL} -)
same = n,Playback(welcome)
*same = n,Dial(DAHDI/g0/88,30,M(outgoing_connect,s,1)^S(30))*
same = n,hangup
exten = h,1,NoOP(Call hangup === outgoing_ivrs context)
same = n,NoOP(calldate2=${CDR(calldate)},src2=${CDR(src)},dst2=${CDR(dst)})
same =
n,NoOP(channel2=${CDR(channel)},dstchannel2=${CDR(dstchannel)}start2=${CDR(start)})
same =
n,NoOP(end2=${CDR(end)},duration2=${CDR(duration)},billsec=${CDR(billsec)},disposition2=${CDR(disposition)})
[macro-outgoing_connect]
exten = s,1,NoOP( === outgoing_connect ===)
same = n,Playback(welcome_friend)
Thanks in advance. please help out to achieve this requirement.
--
Best Regards,
Rajni Vanza
Consultant Technology
---
Working On Linux,C/C++,VoIP Technology
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[asterisk-users] Transfer call question
Hello guys, I have trunk “1, Internal num “99 and MeetMe “1010 now I calling 99 - 89264959635 via 1 /pbx/agi.php: [agi_channel] = PJSIP/99-0012 /pbx/agi.php: [agi_callerid] = 99 /pbx/agi.php: [agi_calleridname] = 99 /pbx/agi.php: [agi_context] = dialmap /pbx/agi.php: [agi_extension] = 89264959635 then I would like to direct transfer this call to 1010 and when I do that from my phone I getting this agi_request in AGI: /pbx/agi.php: [agi_channel] = PJSIP/1-0013 /pbx/agi.php: [agi_callerid] = 89264959635 /pbx/agi.php: [agi_calleridname] = unknown /pbx/agi.php: [agi_context] = dialmap /pbx/agi.php: [agi_extension] = 1010 There is no information who is transferring that call, so AGI thinks that it is inbound call and hangup it because in my case external 89264959635 to internal 1010 is denied. is there way do determine that call was transfered from 99 so I can use route table of abonent 99 to connect the call properly? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get 2 CDR Records of 2 outgoing calls bridge
Hi All,
I need 2 CDR Records of below 2 numbers for outgoing calls, detail is given
as below:
*9688*
*= Call file : outbound call generate through below file*
Test.call
==
Channel: local/s@outgoing/n
WaitTime: 45
Context: outgoing_ivrs
Extension: s
Priority: 1
Set: contact_no=96
extensions.conf
[outgoing]
exten = s,1,NoOP(- First LEG CALL from call file for outgoing:
Channel-${CHANNEL} -)
*same = n,Dial(DAHDI/g0/${contact_no},45)*
same = n,Noop(* DIALSTATUS-${DIALSTATUS}*)
same = n,hangup
exten = h,1,NoOP(Call hangup === outgoing context)
same = n,NoOP(calldate2=${CDR(
calldate)},src2=${CDR(src)},dst2=${CDR(dst)})
same =
n,NoOP(channel2=${CDR(channel)},dstchannel2=${CDR(dstchannel)}start2=${CDR(start)})
same =
n,NoOP(end2=${CDR(end)},duration2=${CDR(duration)},billsec=${CDR(billsec)},disposition2=${CDR(disposition)})
[outgoing_ivrs]
exten = s,1,Noop(- Second LEG CALL from call file for outgoing_ivrs:
Channel-${CHANNEL} -)
same = n,Playback(welcome)
*same = n,Dial(DAHDI/g0/88,30,M(outgoing_connect,s,1)^S(30))*
same = n,hangup
exten = h,1,NoOP(Call hangup === outgoing_ivrs context)
same = n,NoOP(calldate2=${CDR(calldate)},src2=${CDR(src)},dst2=${CDR(dst)})
same =
n,NoOP(channel2=${CDR(channel)},dstchannel2=${CDR(dstchannel)}start2=${CDR(start)})
same =
n,NoOP(end2=${CDR(end)},duration2=${CDR(duration)},billsec=${CDR(billsec)},disposition2=${CDR(disposition)})
[macro-outgoing_connect]
exten = s,1,NoOP( === outgoing_connect ===)
same = n,Playback(welcome_friend)
Thanks in advance. please help out to achieve this requirement.
--
Best Regards,
Rajni Vanza
Consultant Technology
---
Working On Linux,C/C++,VoIP Technology
Mumbai
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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Hello, I noticed something that might be a result from the fix suggested here, so I'll continue a bit on this thread. After removing the callbackextension field from my realtime sip peer table, the following started happening: I issued command 'sip reload' on the cli and get the following warning: WARNING[24427]: res_config_mysql.c:501 realtime_multi_mysql: MySQL RealTime: Failed to query database: Unknown column 'callbackextension' in 'where clause' This must be a result from removing that field from the db, but somewhere in the code there is a select statement where the callbackextension field is used in the where clause, resulting to the above warning. I wonder if this something to be worried about, or is going to cause problems later? My goal is of coure just to handle calls, save cdrs, do pbx features etc with this asterisk. cheers, Olli 2014-07-15 16:56 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com: Wow, thanks Joshua, it would've taken me forever to find the answer there. It did the trick and the registrations look much better. Merci beaucoup! - Olli 2014-07-15 16:26 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that had no effect on the outcome. The callbackextension is the reason this is happening. From sip.conf.sample: ; A similar effect can be achieved by adding a callbackextension option in a peer section. ; this is equivalent to having the following line in the general section: ; ;register = username:secret@host/callbackextension ; ; and more readable because you don't have to write the parameters in two places ; (note that the port is ignored - this is a bug that should be fixed). Remove that column from your table, restart Asterisk, and it should go away. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Ring
I have this working but I have one problem. I need to grab values from
variables that I have set in the calling context to dial. How would I do that.
[tbs-utils]
exten = s,1,NoOp(Entering tbs-utils for extension ${ARG1})
;Set local variables to be used in the call
same = n,Set(NUMBER=${ARG1})
same = n,Set(GLOBAL(DIALGROUP1)=)
same = n,Set(GLOBAL(DIALGROUP2)=)
same = n,Set(_VM=)
same = n,Set(_TIMER1=)
same = n,Set(_TIMER2=)
same = n,Set(BRANCH=)
same = n,Set(_TO_VM=0)
;Check to see if the Primary SIP trunk is up
same = n,Set(NETWORKSTATUS=${SIPPEER(${GLOBAL(TRUNK1)},status)})
;Setting the TRUNK variable based upon the status of whether Trunk1 is reachable
same =
n,Set(TRUNK=${IF($[$[NETWORKSTATUS=UNREACHABLE]]?${GLOBAL(TRUNK2)}:${GLOBAL(TRUNK1)})})
;Calling the agi script
same = n,AGI(agi://localhost/tbs.agi)
;Displaying the values of the variables set in the agi script
same = n,NoOp(Branch number is: ${BRANCH})
same = n,NoOp(DIALGROUP1 is: ${DIALGROUP1})
same = n,NoOp(DIALGROUP2 is: ${DIALGROUP2})
same = n,NoOp(TIMER1 is: ${TIMER1})
same = n,NoOp(TIMER2 is: ${TIMER2})
same = n,NoOp(VM is: ${VM})
same = n,NoOp(TO_VM is: ${TO_VM})
;Check to see if we should go straight to VM
same = n,Gotoif($[${TO_VM} = 1]?200:)
;Dial the primary number and to to the return status
same = n,Dial(Local/Group1-101@DelayLocal/Group2-101@Delay,30)
same = n,Hangup();
[Delay]
;Dial Group 1
exten = Group1-101,1,Verbose(2,Dialing Group 1 set of phones
${GLOBAL(DIALGROUP1)})
same = n,Dial(${DIALGROUP1},20,t)
;Dial Group 2
exten = Group2-101,1,Verbose(2,Dialing Group2 set of phones)
same = n,Verbose(2, Waiting 10 seconds before dialing)
same = n,Wait(10)
same = n,Dial(${DIALGROUP2},${TIMER2},t)
Thanks,
Scott Haley
5-2244
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Thursday, July 17, 2014 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
Thanks AJ, this sounds like what I need.
Thanks,
Scott Haley
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Thursday, July 17, 2014 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
On Wednesday 16 Jul 2014, Haley,Scott A wrote:
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial
another set of numbers while continuing to ring, or interrupting the
first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
This sounds exactly like the sort of situation for which local channels were
invented .
Dial(${DIALGROUP1}LOCAL/foo@bar) with a longer timeout than 10 seconds. Then
in your local channel, wait 10 and Dial(${DIALGROUP2}). The first Dial() will
be satisfied when someone answers either a phone in dial group 1, or a phone in
dial group 2 set ringing by the Dial() in the local channel.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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[asterisk-users] chan_motify / res_xmpp bind address?
I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Ring
On Friday 18 Jul 2014, Haley,Scott A wrote: I have this working but I have one problem. I need to grab values from variables that I have set in the calling context to dial. How would I do that. I think you need to prefix your variable names with *two* underscores, to make them indefinitely heritable down the succession of channels. If they are prefixed with a single underscore, then they only get inherited *once*; so if the child channel spawns a grandchild, then any _VARS it inherited from the parent channel won't exist in the grandchild, but any __VARS will. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Ring
That worked. I had to use the *two* underscores in the agi script where I was setting the values. Thanks. Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, July 18, 2014 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simultaneous Ring On Friday 18 Jul 2014, Haley,Scott A wrote: I have this working but I have one problem. I need to grab values from variables that I have set in the calling context to dial. How would I do that. I think you need to prefix your variable names with *two* underscores, to make them indefinitely heritable down the succession of channels. If they are prefixed with a single underscore, then they only get inherited *once*; so if the child channel spawns a grandchild, then any _VARS it inherited from the parent channel won't exist in the grandchild, but any __VARS will. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Ring
On Friday 18 Jul 2014, Haley,Scott A wrote: That worked. I had to use the *two* underscores in the agi script where I was setting the values. Thanks. Glad you got it working in the end! I always like to use plenty of NoOp() statements to make sure the variables I'm setting are correct, especially when it requires fiendish logic which has to jump around between contexts and spawn channels. If you are in the habit of using n for next instead of explicit step numbers, so much the better, as this means extraneous NoOp()s can easily be commented out or removed later. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP over 3G/4G Data
What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff’s breaking the connection. This will be for fixed wireless. I have read to use G.729 codec, and TCP for signaling to bypass firewalls. Besides that, what other settings are recommended? Changes in MTU? Changes in voice payload ms? Is there a better codec to use? Header compression? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over 3G/4G Data
Depends on the carrier. Verizon Wireless appears to activly block SIP. G729 codec is needed on 3G and is a good idea on 4G. I use TLS and SRTP to work around carrier stupidity. I also use a non-standard port for TLS. It mostly works much of the time. Don’t get BRIA, every time your registration is dropped it will popup a notification. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Dude Sent: Friday, July 18, 2014 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoIP over 3G/4G Data What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff’s breaking the connection. This will be for fixed wireless. I have read to use G.729 codec, and TCP for signaling to bypass firewalls. Besides that, what other settings are recommended? Changes in MTU? Changes in voice payload ms? Is there a better codec to use? Header compression? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over 3G/4G Data
On 18-07-14 17:59, Tech Dude wrote: What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff’s breaking the connection. This will be for fixed wireless. I have read to use G.729 codec, and TCP for signaling to bypass firewalls. Besides that, what other settings are recommended? Changes in MTU? Changes in voice payload ms? Is there a better codec to use? Header compression? Use TLS/SRTP so the carrier can't do packet inspection/snooping and mess with or block your VoIP connections. They might throttle/block port 5060 and 5061 anyway in which case you should use different ports. I use Asterisk 11 with TLS/SRTP, G.722 and Android phones (4G, CSipSimple or Bria), iPhones (4G, Bria) and Polycom phones. G.722 on inter-office calls has been working great so no need for G.729 and Asterisk uses standard alaw/ulaw for regular PSTN calls. I use TCP, have made no MTU changes and use standard 20ms voice payload. Packet loss, latency and jitter are the enemy. Your router might be top notch but if the cell tower is overloaded and experiences too much packet loss, delays of more than 150ms and lots of jitter than you may get crappy sound quality. If possible, get some prepaid 4G sim cards for your router from different carriers and test which carrier consistently provides the best signal, least delay, packet loss and jitter. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over 3G/4G Data
We use standard sip ports all day long, and have had no issues with employee phones on the verzion network. On Jul 18, 2014 12:03 PM, Eric Wieling ewiel...@nyigc.com wrote: Depends on the carrier. Verizon Wireless appears to activly block SIP.G729 codec is needed on 3G and is a good idea on 4G. I use TLS and SRTP to work around carrier stupidity. I also use a non-standard port for TLS. It mostly works much of the time. Don’t get BRIA, every time your registration is dropped it will popup a notification. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tech Dude *Sent:* Friday, July 18, 2014 1:00 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] VoIP over 3G/4G Data What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff’s breaking the connection. This will be for fixed wireless. I have read to use G.729 codec, and TCP for signaling to bypass firewalls. Besides that, what other settings are recommended? Changes in MTU? Changes in voice payload ms? Is there a better codec to use? Header compression? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
