[asterisk-users] Regarding XRMS support
Hi, I installed the XRMS Open source CRM(xrms-2006-07-25-v1.99.2-.tar.gz) in the PC. Installed the CTI plugin inside as per the readme file inside /plugins/cti/README.txt(I am attaching that as well for the reference). I could not able to view the later development inside the application. Can anyone please share any idea how to approach for the configuration of *Asterisk CTI with XRMS in the Windows* environment. as in the readme file of that package it's it was under active development.. As I am new to this It will be of great help for me to get atleast the steps or any kind of source for that. Thanks in advance. Asit CTI / Asterisk Out Dial XRMS Plugin v0.1 uses the Asterisk Open Source Soft PBX from: http://www.asterisk.org/ copyright 2004 Glenn Powers gl...@net127.com Licensed Under the Open Software License v. 2.0 *** THIS PLUGIN IS UNDER ACTIVE DEVELOPMENT *** YOU *MUST* INSTALL THE TABLE IN cti-call-queue.sql to your database BEFORE you activate this plugin. If you don't, you'll get a Javascript error every second on every page. The screen pop function uses the JPSpan library. (It's one of those cool Ajax things you've been hearing about.) The Cisco 7960 config writer currently provides only basic functionality. Specifically, you can't edit a config file with XRMS once you create it. XRMS Voice Mail Plugin v0.1 for use with the Asterisk Open Source PBX http://www.asterisk.org/ see voicemail-install.txt The XRMS username is used to look up the correct extension in /etc/asterisk/voicemail.conf If the XRMS username is not in voicemail.conf, this plugin will not show any messages. The *last* match is used. This is a hack. Patches welcomed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] dialplan tips
2009/8/6 Alex Samad a...@samad.com.au
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote:
Here's how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten = 101,n,Playback(vm-goodbye)
exten = 101,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
exten = 101-BUSY,n,Wait(3)
exten = 101-BUSY,n,VoiceMail(1...@default,b)
exten = 101-BUSY,n,Playback(vm-goodbye)
exten = 101-BUSY,n,Hangup()
I have a question about this dialplan, why does the dial do a jump to
101-DIALSTATUS, is there a goto 101-DIALSTATUS missing ?
Alex
Hi Alex
You are right, a line is missing. If I just stay in this same dialplan, I
will add this line
exten = 101,GotoIf([${DIALSTATUS} = BUSY]?101-BUSY,1)
just after the diap() application.
But as someone else wrote before, you can do a dialplan like this
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten = 101-NOANSWER,n,Playback(vm-goodbye)
exten = 101-NOANSWER,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
exten = 101-BUSY,n,Wait(3)
exten = 101-BUSY,n,VoiceMail(1...@default,b)
exten = 101-BUSY,n,Playback(vm-goodbye)
exten = 101-BUSY,n,Hangup()
exten = _101-.,1,Goto(101-NOANSWER,1)
Harry
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Re: [asterisk-users] sip.conf parameter and sip msg between server - client
- what's the difference between a subscribe request et a register request ? A subscription in the SIP protocol is saying Hey, I'd like to be notified when something happens. This is most often used when a phone wants to subscribe to the state of another extension, or to the status of a voicemail box. A registration is where one SIP device tells another Hey, I'm over here. If you get any calls for me, send them to me at this IP address and port. -- Jared Smith Training Manager Digium, Inc. Hi Jared Thx for your explanation. So in sip.conf if I set allowsubscribe to no in [general] section that will mean by default all my device I will declare will no be able to receive SUBSCRIPTION msg ? Except if I declare allowsubscribe for one of them... now I have a better understanding of what a subscription is, these questions are coming about some parameter in sip.conf : - 'notifyringing' set to 'yes' will send sorry but I'm already on call to any device when my device named john is already in use ? Is it an issue to limit incoming/outgoing call for a device ? - 'callcounter' is it an issue to limit incoming/outgoing call for a device ? - I read that 'call-limit' or 'busylevel' (in asterisk 1.6) is an issue to limit incoming/outgoing call on a device. Is it true ? Can I use them together or do I have to use just one ? Regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
2009/8/6 Alex Balashov abalas...@evaristesys.com Sure it is. Just get a media gateway that does T.38 - and does it relatively well. Wich the Pattons do quite well afaik. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] 20 seconds cut off problem
Hi I'm having an issue with just one of the phones (snom300) attached to our asterisk server (1.4.17 using RealTime) Sometimes (not consistently), any outbound call cust off at 20 seconds exactly and I see the following in my asterisk console [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1946 retrans_pkt: Maximum retries exceeded on transmission 3c3251e0edaf-4jnjbmy9uupi for seqno 2 (Critical Response) [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1970 retrans_pkt: Hanging up call 3c3251e0edaf-4jnjbmy9uupi - no reply to our critical packet. There as another SIP phone plugged into the same router and that has no issues at all and inbound calls are not affected either. The codecs order on the phone match up to those set on the server (g729;alaw;ulaw;;). There are about 50 other phones attached to the server and none of the others have this issue. Well actually, one did but that person got a new handset (they were previously using a very old and rubbish Grandstream) and the problem immediately stopped. Has anyone experienced anything like this before? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] dialplan tips
harry R schrieb:
But as someone else wrote before, you can do a dialplan like this
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten = 101-NOANSWER,n,Playback(vm-goodbye)
exten = 101-NOANSWER,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
exten = 101-BUSY,n,Wait(3)
exten = 101-BUSY,n,VoiceMail(1...@default,b)
exten = 101-BUSY,n,Playback(vm-goodbye)
exten = 101-BUSY,n,Hangup()
exten = _101-.,1,Goto(101-NOANSWER,1)
Such a dialplan is potentially dangerous.
Someone could call 101-BUSY directly and leave a voicemail message
even if you are available.
Please don't jump to extensions. Use labels (priorities) instead.
Or use the switch(){} statement in AEL (:-) so you don't have to
worry about the implementation details.
Philipp Kempgen
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[asterisk-users] queue agents get stuck
Hello to all I have a queue where often my agents get stuck and cannot logoff. This is very bad, because agents cannot login again, and in Queuemetrics reports the agents appear to be online. How can I create a timeout to my agents and for the queue to kick them? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] E1 line simulation for Asterisk
Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] ntop and Asterisk
Hi, Would it be sane to run ntop on the same box as Asterisk or better to mirror a LAN port etc? http://www.ntop.org/OpenSourceVoipMonitoring.pdf Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] dialplan tips
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten = 101-NOANSWER,n,Playback(vm-goodbye)
exten = 101-NOANSWER,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
exten = 101-BUSY,n,Wait(3)
exten = 101-BUSY,n,VoiceMail(1...@default,b)
exten = 101-BUSY,n,Playback(vm-goodbye)
exten = 101-BUSY,n,Hangup()
exten = _101-.,1,Goto(101-NOANSWER,1)
Such a dialplan is potentially dangerous.
Someone could call 101-BUSY directly and leave a voicemail message
even if you are available.
Please don't jump to extensions. Use labels (priorities) instead.
Or use the switch(){} statement in AEL (:-) so you don't have to
worry about the implementation details.
Philipp Kempgen
Hi Philipp
How could someone call 101-BUSY directly ? I dont see a way to that...
But if I wrote something like
exten = 101,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,VoiceMail(1...@default,u)
exten = s-BUSY,n,VoiceMail(1...@default,b)
exten = _s-.,1,Goto(101-NOANSWER,1)
there is a problem to ???
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Re: [asterisk-users] Best ISDN BRI solutions?
Moises Silva moises.si...@gmail.com writes: Just for the record, Sangoma Media Gateway does exactly that, leave all your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with Asterisk through the Woomera protocol. What is the advantage of doing Woomera rather than a complete SIP gateway? Is Woomera hardware that much cheaper than e.g. SIP-to-PRI? Is it easier to configure? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 - 1.4 CDR change?
Greetings list,
Wondering if anyone can shed some light on an apparent change in CDRs between
1.2 and 1.4.
One of our clients runs a virtual PA service and has a few hundred DDIs - one
for each client. Their * box is set up with short codes they prefix their calls
with to set the correct outbound CLID, so for example:
123*number_to_be_dialled
This simply does the following:
Set(CALLERID(number)=clid_to_use
Goto(outbound,${EXTEN:4},1)
In * 1.2, this resulted in the callerid field of their CDRs reflecting the
change from their extension number to clid_to_use, which they could then use
to correctly apportion outbound calls to their clients. However, in 1.4, the
callerid field retains the original extension number which initiated the call,
which means billing is next to impossible based on callerid.
Our current workaround is to append the CLID to the accountcode field, which
works, but does require extra processing of the CDRs each month, which is far
from ideal.
So, two questions if I may:
1) is this a result of the 1.2 - 1.4 upgrade, or is it possible something else
has caused this? (basically, is this a known issue or feature of 1.4?)
2) Is it possible to revert back to the previous behaviour?
Thanks in advance!
Regards,
Chris
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Re: [asterisk-users] E1 line simulation for Asterisk
http://kb.digium.com/entry/95/ On Thu, Aug 6, 2009 at 6:02 AM, ABBAS SHAKEELshakeel.abbas@gmail.com wrote: Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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_mobile handle 92 log flood
Dear all, Picked up some more BT usb adapters and got a flood of error messages as follows: hci_scodata_packet: *hci0 SCO packet for unknown connection handle 92* Anyone has any idea how to deal with this? Sasa Bobek ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Can't delete voicemail messages
Hello, I'm running Asterisk 1.6.1.0 compiled from scratch under debian Lenny and I can't delete message from VoiceMailMain using option 7 Default folder is /var/spool/asterisk/voicemail and it's owned by asterisk:asterisk with 777 permissions Apparently VoicemailMain delete the message and inmediatly undelete it ! This the same issue as in this post : http://www.mail-archive.com/asterisk-users@lists.digium.com/msg198336.html Note that I'm using the spanish voiceset. == Parsing '/var/spool/asterisk/voicemail/from-internal/300/Old/msg.txt': == Found -- Playing '/var/spool/asterisk/voicemail/from-internal/300/Old/msg.slin' (language 'es') -- Playing 'vm-deleted.alaw' (language 'es') -- Playing 'vm-undeleted.alaw' (language 'es') -- Playing 'vm-advopts.alaw' (language 'es') == Spawn extension (from-telebullas, *98, 3) exited non-zero on 'SIP/300-6d What I've missed ? Bes regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] 20 seconds cut off problem
Hello. I think i've seen this problem, it was generated by a missing ACK on 200 OK. If that is the case try modifying session timer parameters in sip.conf so a missing ACK will not lead to call termination. Peter Ishfaq Malik wrote: Hi I'm having an issue with just one of the phones (snom300) attached to our asterisk server (1.4.17 using RealTime) Sometimes (not consistently), any outbound call cust off at 20 seconds exactly and I see the following in my asterisk console [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1946 retrans_pkt: Maximum retries exceeded on transmission 3c3251e0edaf-4jnjbmy9uupi for seqno 2 (Critical Response) [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1970 retrans_pkt: Hanging up call 3c3251e0edaf-4jnjbmy9uupi - no reply to our critical packet. There as another SIP phone plugged into the same router and that has no issues at all and inbound calls are not affected either. The codecs order on the phone match up to those set on the server (g729;alaw;ulaw;;). There are about 50 other phones attached to the server and none of the others have this issue. Well actually, one did but that person got a new handset (they were previously using a very old and rubbish Grandstream) and the problem immediately stopped. Has anyone experienced anything like this before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] E1 line simulation for Asterisk
thanks alot David that was really helping how can i test for calls etc ie to generate and make calls stuff like that On Thu, Aug 6, 2009 at 3:30 PM, David Backebergdbackeb...@gmail.com wrote: http://kb.digium.com/entry/95/ On Thu, Aug 6, 2009 at 6:02 AM, ABBAS SHAKEELshakeel.abbas@gmail.com wrote: Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] 20 seconds cut off problem
Cheers for that. We solved the problem this time by updating the routers firmware to the latest version but if it happens again I'll look into what you're saying. Can I just clarify that you mean the RTP Timers though? It's a production server and doing sip reloads is massively frowned upon. Thanks again Ish Peter Johansson wrote: Hello. I think i've seen this problem, it was generated by a missing ACK on 200 OK. If that is the case try modifying session timer parameters in sip.conf so a missing ACK will not lead to call termination. Peter Ishfaq Malik wrote: Hi I'm having an issue with just one of the phones (snom300) attached to our asterisk server (1.4.17 using RealTime) Sometimes (not consistently), any outbound call cust off at 20 seconds exactly and I see the following in my asterisk console [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1946 retrans_pkt: Maximum retries exceeded on transmission 3c3251e0edaf-4jnjbmy9uupi for seqno 2 (Critical Response) [Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1970 retrans_pkt: Hanging up call 3c3251e0edaf-4jnjbmy9uupi - no reply to our critical packet. There as another SIP phone plugged into the same router and that has no issues at all and inbound calls are not affected either. The codecs order on the phone match up to those set on the server (g729;alaw;ulaw;;). There are about 50 other phones attached to the server and none of the others have this issue. Well actually, one did but that person got a new handset (they were previously using a very old and rubbish Grandstream) and the problem immediately stopped. Has anyone experienced anything like this before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] E1 line simulation for Asterisk
On Thu, 6 Aug 2009, ABBAS SHAKEEL wrote: thanks alot David that was really helping how can i test for calls etc ie to generate and make calls stuff like that I think what Dvid didn't say was that you need a 2nd Asterisk box with E1 card, as well as that cable Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] E1 line simulation for Asterisk
Lolz exactly right thats what i was wondering with TDM400P can i do that i think i need TE420 for this . I will get that after two or three days then i will ask later on how to test thanks Alot On Thu, Aug 6, 2009 at 5:14 PM, Gordon Hendersongordon+aster...@drogon.net wrote: On Thu, 6 Aug 2009, ABBAS SHAKEEL wrote: thanks alot David that was really helping how can i test for calls etc ie to generate and make calls stuff like that I think what Dvid didn't say was that you need a 2nd Asterisk box with E1 card, as well as that cable Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 dont detects hangup by phone
Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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't delete voicemail messages
On 6 Aug 2009, at 11:33, Sébastien Cramatte wrote: Apparently VoicemailMain delete the message and inmediatly undelete it ! This the same issue as in this post : http://www.mail-archive.com/asterisk-users@lists.digium.com/msg198336.html Note that I'm using the spanish voiceset. == Parsing '/var/spool/asterisk/voicemail/from-internal/300/Old/ msg.txt': == Found -- SIP/300-6d2b0eb8 Playing '/var/spool/asterisk/voicemail/ from-internal/300/Old/msg.slin' (language 'es') -- SIP/300-6d2b0eb8 Playing 'vm-deleted.alaw' (language 'es') -- SIP/300-6d2b0eb8 Playing 'vm-undeleted.alaw' (language 'es') -- SIP/300-6d2b0eb8 Playing 'vm-advopts.alaw' (language 'es') == Spawn extension (from-telebullas, *98, 3) exited non-zero on 'SIP/300-6d What I've missed ? Try debugging DTMF, it could be reading the '7' twice. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 don't detects hang-up by phone
Assuming you are connected to a regular phone line, the hang up signal from the phone line would be a break or reversal of polarity of the DC signal on the phone line. (We connect to PRIs, so our signaling is on a data channel. I assume you don't. ) The first question you need to answer is Are you getting a voltage drop or polarity reversal when the other end disconnects? Asterisk has to have a signal to respond to. Some Telcos may not give that signal. Check your phone line with a meter. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL Sent: Thursday, August 06, 2009 7:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk dont detects hangup by phone Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 don't detects hang-up by phone
I was having the same problem with about half of my POTS lines. I switched the polarity on the connections for those lines and the problem disappeared. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Thursday, August 06, 2009 9:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk don't detects hang-up by phone Assuming you are connected to a regular phone line, the hang up signal from the phone line would be a break or reversal of polarity of the DC signal on the phone line. (We connect to PRIs, so our signaling is on a data channel. I assume you don't. ) The first question you need to answer is Are you getting a voltage drop or polarity reversal when the other end disconnects? Asterisk has to have a signal to respond to. Some Telcos may not give that signal. Check your phone line with a meter. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL Sent: Thursday, August 06, 2009 7:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk dont detects hangup by phone Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Set PHP binary location for AGI
I am not finding anything relating to this on Google, so I thought I'd pose the question here... I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a custom built PHP5.2.10 install to interconnect with our Firebird SQL database, which I've done. But I noticed that the default install path for PHP5 on this box appears to be /usr/local/bin/php rather than the path that the default PHP5.16 path of /usr/bin/php. To be certain that the correct PHP binaries are being called, is there a conf setting somewhere that I can tell Asterisk AGI where the PHP binary is that I want to use for this? I noticed that most AGI PHP scripts begin with: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q for my build, but is this enough? Is there somewhere else that needs to be updated in order for AGI to correctly find the PHP build I've done? Myles P.S. I did try and change the PHP configure options to install in /usr/bin but that didn't work because it won't install the man pages and headers in there for some reason. But its installing fine in /usr/local/bin which appears to be its default install location anyway, so I'd prefer to go with that and keep this simple if I can. -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Monitoring Asterisk uptime
We have added Asterisk to a line of 'mission critical' servers at our business, and being in the web application development business one of the core things we do is to monitor web server availability. I'd like to add Asterisk to the servers that our monitoring systems are handling, and also that our SIP trunk provider has our Asterisk system correctly registered at all times. What are the 'best practice' tricks used for monitoring an Asterisk phone system for uptime and SIP registration from an external monitoring server? I can certainly ping the box, but I really need more than this. I need to know if the Asterisk service is running, and also that there hasn't been any issues with SIP registration to our external trunks. If anyone could share how they are doing this sort of thing, it would be greatly appreciated. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Set PHP binary location for AGI
On 6 Aug 2009, at 15:21, Myles Wakeham wrote: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q for my build, but is this enough? You could always test it.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Set PHP binary location for AGI
On Thu, 6 Aug 2009, Myles Wakeham wrote: I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a custom built PHP5.2.10 install to interconnect with our Firebird SQL database, which I've done. But I noticed that the default install path for PHP5 on this box appears to be /usr/local/bin/php rather than the path that the default PHP5.16 path of /usr/bin/php. To be certain that the correct PHP binaries are being called, is there a conf setting somewhere that I can tell Asterisk AGI where the PHP binary is that I want to use for this? I noticed that most AGI PHP scripts begin with: There is no conf setting. Asterisk does not know nor care what language you write your AGI in, as long as you follow the AGI protocol. #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q This should work. Did you try it? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Monitoring Asterisk uptime
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need host keys to do
this without user interation), you should be able to poll any linux server for
really anything you want.
Someone who is in the business of selling hosted applications should be able to
EASILY use awk and grep to figure out if his sip trunks are in the 'Registered'
state via SSH... Unless I misunderstood the nature of your question and you
were looking for something native to asterisk or the AMI.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham
Sent: Thursday, August 06, 2009 10:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitoring Asterisk uptime
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server availability.
I'd like to add Asterisk to the servers that our monitoring systems are
handling, and also that our SIP trunk provider has our Asterisk system
correctly registered at all times.
What are the 'best practice' tricks used for monitoring an Asterisk
phone system for uptime and SIP registration from an external monitoring
server? I can certainly ping the box, but I really need more than this.
I need to know if the Asterisk service is running, and also that there
hasn't been any issues with SIP registration to our external trunks.
If anyone could share how they are doing this sort of thing, it would be
greatly appreciated.
Thanks
Myles
--
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
http://www.techsolusa.com
Phone +1-480-451-7440
___
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[asterisk-users] Setting up Outgoing Trunk
Hello Everybody, I have a genuine problem in Asterisk setup. I have three inbound trunks in my asterisk box, everything is working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should get different no and if extn: 202 make an outgoing call the recipient should get different one. Please can someone help me in this. Thanks Shantanu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Setting up Outgoing Trunk
Are these trunks or PRI/ISDN circuits, or phone lines? If either of the first two, the callerID sent with the call should be their ID, which should be the appropriate number of digits your area telco expects. Depending on your agreement with them, they may be supplying the number, rather than accept what you send. If your connection is phone lines they are supplying the Line Number, and you have no control over that except by strategic use of the lines, etc. Or if there is further info or questions, explain the exact details. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kumarshantanu Sent: Thursday, August 06, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Setting up Outgoing Trunk Hello Everybody, I have a genuine problem in Asterisk setup. I have three inbound trunks in my asterisk box, everything is working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should get different no and if extn: 202 make an outgoing call the recipient should get different one. Please can someone help me in this. Thanks Shantanu http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/sign atureline@middle? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] ForkCDR and setting the account info?
I've been Googling all morning and searching voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a how to. I'd like to then assign specific accounts in the CDRs. Possible? Thanks, PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Setting up Outgoing Trunk
On 6 Aug 2009, at 16:32, kumarshantanu wrote: Hello Everybody, Hi. I have a genuine problem in Asterisk setup. Ok. I have three inbound trunks in my asterisk box, everything is What kind of trunks. working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Ok. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should get different no and if extn: 202 make an outgoing call the recipient should get different one. Ok. Please can someone help me in this. If you show us some config, tell us trunk types and generally 'giving us something to go on. Thanks Shantanu Steve Heh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calls Disconnecting out of the blue .. [Renamed]
Greetings again List. I'm facing a strange case with one of the productive Asterisk servers.. i have 3 providers sending traffic to the call center where agents pickup the calls. calls come into the server Queue Agents Last October .. an undersea cable got disconnected placing Egypt and the countries in the region offline.. when internet came back .. the call center located in Egypt had no SIP protocol working.. and we shifted to IAX.. 26 days later SIP started to work again .. but since then calls started to disconnect out of the blue.. we get calls that may last for 45 minutes.. and end normaly .. and we get calls that ring and disconnect the moment the agent picks up been facing a problem with my client as they use the Flash Operator Panel to monitor the call flow through the server and the regualr setup Queue Local users won't work for them as the Flash operator flash offline static agents as online so the client won't know who is on and who is off.. and it's impossible to teach the agents to Login and Logoff the Queue.. so the only solution is the following.. Caller Queue FindMeFollowMe Extension Local SIP extensions this way .. my client is able to monitor the calls and things won't get complicated.. (this is the setup we have been using for 6 months before the problem with the internet occures) since the internet problem and calls are getting disconnected .. out of the blue.. nothing has changed.. and to make sure things are going well .. we moved the server to a Hosting company in California with 10 mb/s connection speed.. (Same Setup that was working well) and still calls get disconnected.. after a lot of problems with the client .. i asked them to change the ISP (my prime suspect was the internet) and finaly they managed to change the ISP .. but the problem is still there.. my server informations are the following Asterisk 1.4.22-3 Uname -a: Linux 2.6.18-92.1.18.el5 sip.conf ;;Agent Sample from Sip.conf [3000] type=friend secret=3000 qualify=yes port=5060 disallow=all allow=g729 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/3000 context=from-internal canreinvite=no call-limit=1 busy-limit=1 ;;Provider's Sample from Sip.conf [50011] type=peer qualify=yes port=5060 pickupgroup= nat=no host=XXX.YYY.ZZZ.NNN disallow=all allaw=alaw allaw=ulaw allow=g729 dial=SIP/50011 context=from-internal canreinvite=no deny=0.0.0.0/0.0.0.0 permit=XXX.YYY.ZZZ.NNN/255.255.255.255 # extensions.conf ;;the provider sends calls to Virtual DIDs (Extensions) in my system which is 8000 exten = 8000,1,GotoIfTime(07:00-16:00|sat-fri|1-31|jan-dec?ext-queues,*8000,1) exten = 8000,n,Answer exten = 8000,n,Queue(8000,t,,,10) exten = 8000,n,Dial(IAX2/6005:6...@backupserver/11) ;; sends the call to a backup server. exten = *8000,1,Answer exten = *8000,n,Dial(IAX2/6005:6...@backupserver/11) the Providers strictly send calls with codec G.729 my agents get best voice quality with G.711u I need your advice .. am i missing anything in this setup?? it used to work .. and it STILL works on another hosted server with Agents located in Morocco.. with a different version of Asterisk 1.4.20-1 and better hold time for the calls.. -- AHD Tarek Sawah _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Monitoring Asterisk uptime
David writes:
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep
USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need host
keys to do this without user interation), you should be able to poll
any linux server for really anything you want.
Someone who is in the business of selling hosted applications should
be able to EASILY use awk and grep to figure out if his sip trunks are
in the 'Registered' state via SSH... Unless I misunderstood the nature
of your question and you were looking for something native to asterisk
or the AMI.
Thanks for the feedback, and its a good idea! We are monitoring about
45 servers right now with some 3rd party monitoring software, and
looking to move it all to Nagios. Unfortunately our currently software
doesn't give me as much flexibility on a per device basis to have it
execute a script for each device that its monitoring to this level, but
I'm thinking that we could customize Nagios to do this sort of thing.
Its definitely worth exploring and I'll post back with the results.
Thanks again.
Myles
--
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
http://www.techsolusa.com
Phone +1-480-451-7440
___
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Register Now: http://www.astricon.net
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Re: [asterisk-users] Set PHP binary location for AGI
Steve writes: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q This should work. Did you try it? Yes, its working fine. The only problem is that when I went to 'uninstall' the standard PHP installation that came with CentOS 5 on this box, it de-registered PHP from executing correctly but left all the old files still in place. So when I re-installed a new PHP build, I can't see what version is actually running from AGI. I'll get AGI to do a phpinfo() call and see if I can pipe the results to a file so I can see what is going on here. From what I can tell, however, it would appear that the php executable at /usr/local/bin is running I was just hopeful that a conf setting somewhere could tell AGI what language/host is to be used for executing its calls. But if I can rely on the #! setting in the file, that's good enough for me. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inbound Call coding
Hi, how can I make an inbound call to be coded by the agent and be stored in the database for a later report?? I mean, I want the agent to dial after the call is finished, a code that means Age information, another code that means where to get more info, etc. I really really appreciate your help. RODRIGO CRUZ E-MAIL. mailto:rodrigo.c...@vcip.com.mx rodrigo.c...@vcip.com.mx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inbound Call coding
You want a dialplan or AGI that works in the h (hangup) context. What you really will probably want is a callback to accept the digits. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Cruz Sent: Thursday, August 06, 2009 12:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Inbound Call coding Hi, how can I make an inbound call to be coded by the agent and be stored in the database for a later report?? I mean, I want the agent to dial after the call is finished, a code that means Age information, another code that means where to get more info, etc. I really really appreciate your help. RODRIGO CRUZ E-MAIL. mailto:rodrigo.c...@vcip.com.mx rodrigo.c...@vcip.com.mx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] No audio on remote SIP calls
Hi everyone. We have an asterisk server in our main office and phones at each remote site. The remote offices are connected via a MPLS which, to my knowledge has no natting going on. The problem I have is that any call from a remote phone to a remote phone (even on the same remote lan) results in no audio. If I make a call from the same LAN the asterisk server is on, to one of these remote sites, I get perfect two way audio. If I play a call from one phone to another at a remote site, there is no audio, however, I do hear messages (such as voicemail, things from Playback(), etc) that originate on the asterisk server. I've tried adjusting canreinvite= in sip.conf in hopes in might have some effect, but so far nothing. Suggestions on where else to look, or what the problem might be? Which configs would be useful in troubleshooting? Thanks. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] ForkCDR and setting the account info?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 J. G. wrote: I've been Googling all morning and searching voip-info.org http://voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a how to. I'd like to then assign specific accounts in the CDRs. Possible? Look up the CDR function. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKeyj/CFu3bIiwtTARAgozAKCPGThtSJzrqs2vlMMvEsfzICDIkgCcD1oE 9oKNSFEDD06hZQ5qa/T9FQI= =YlEU -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 dont detects hangup by phone
At 5:24 PM on 06 Aug 2009, ABBAS SHAKEEL wrote: Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this You may need to enable busydetect in chan_dahdi.conf or zapata.conf. However, if you do enable it, I would recommend you also tune it with the busycount and busypattern options so you don't get false positives (asterisk hanging up in the middle of a call). You can read the comments in the sample configs or on voip-info.org for more information: http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf.sample -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] ForkCDR and setting the account info?
Gah - I've been trying to find the proper search syntax all day.. I Googled asterisk CDR Function and it's the first thing that comes up... Sometimes I wonder whether or not my brain works right.. Thanks Barry! On Thu, Aug 6, 2009 at 3:03 PM, Barry L. Kline blkl...@attglobal.netwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 J. G. wrote: I've been Googling all morning and searching voip-info.org http://voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a how to. I'd like to then assign specific accounts in the CDRs. Possible? Look up the CDR function. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKeyj/CFu3bIiwtTARAgozAKCPGThtSJzrqs2vlMMvEsfzICDIkgCcD1oE 9oKNSFEDD06hZQ5qa/T9FQI= =YlEU -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Extensions patterns algorithm
Hi, has anybody some python code algorithm to parse an extension pattern? I have a number and need to know if match with some pattern. Thansk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inbound Call coding
Danny Nicholas escribió: You want a dialplan or AGI that works in the h (hangup) context. What you really will probably want is a callback to accept the digits. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Cruz *Sent:* Thursday, August 06, 2009 12:58 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Inbound Call coding Hi, how can I make an inbound call to be coded by the agent and be stored in the database for a later report?? I mean, I want the agent to dial after the call is finished, a code that means Age information, another code that means where to get more info, etc... I really really appreciate your help... RODRIGO CRUZ *E-MAIL.** *rodrigo.c...@vcip.com.mx mailto:rodrigo.c...@vcip.com.mx** Or better, make a little web script with a simple form for your agents to capture the information and store it in a database, leaving that job out of the telephony system. That would be far easier, and later you can think of some pop-up CTI with asterisk or something. With all the advantages of web development, the old propietary PBX activity codes seem to be obsolete nowadays. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
On Thu, Aug 6, 2009 at 6:19 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Moises Silva moises.si...@gmail.com writes: Just for the record, Sangoma Media Gateway does exactly that, leave all your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with Asterisk through the Woomera protocol. What is the advantage of doing Woomera rather than a complete SIP gateway? Is Woomera hardware that much cheaper than e.g. SIP-to-PRI? Is it easier to configure? It can be argued that is easier to configure, but I'll leave that to sysadmins to decide. The point is Woomera is not meant to replace SIP or compete with it, is just an easy way to distribute your TDM interfaces and not have them in the same server where you do media processing, IVR, routing etc. This should leave the picture more clear http://wiki.sangoma.com/wanpipe-linux-asterisk-ss7 So typically you will send all your TDM calls to 1 or more Asterisk servers running chan_woomera and any other service you want to provide in there (including SIP outbound calls). -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Voicemail feature: enable or disable the ability to leave a message
Perhaps it's only basic in certain parts of the world... I know I've never experienced a voicemail system with such a feature... I'm not saying having the option would be bad... but... I'd prefer voicemail to get some more common, more requested of me, features first and that's personally where I'd spend my effort... But... feel free to either put a patch together yourself or offer a bounty... d 2009/8/1 Emrah e...@ekanet.net Dom, Quoting myself from your original message: I know everything is possible to be done via the Dialplan. I could just have a Playback to achieve this or pretty easily code my own voicemail app via AGI too. I am not asking how to do this. I know everything is possible with Asterisk. In my opinion, the feature I am talking about is a very basic feature of Voicemail systems and I think it should be natively implemented in Asterisk. Thanks for your hint though. :) Emrah Don Kelly wrote: How 'bout setting up an extension that simply plays an announcement and hangs up. Then transfer calls from extensions that don't want messages to this extension. You could have a few extensions with a few different recordings to suit different situations. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emrah Sent: Friday, July 31, 2009 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail feature: enable or disable the ability to leave a message Mark, I think you did not understand my message. I am accustomed to have the option to allow or disallow the recording of a message in my voicemail, even my mobile carrier provides it. E.g.: I can record a message that says Please call back later, I am currently on the phone. without any beep tone or the possibility to leave a message. I know everything is possible to be done via the Dialplan. I could just have a Playback to achieve this or pretty easily code my own voicemail app via AGI too. I don't catch what you are trying to tell about reading the input in the Dialplan... My scenario is pretty simple, there should be an option in the voicemail to allow the user to choose whether he accepts messages or not. Cheers! Emrah Mark Michelson wrote: Emrah wrote: Hi, I think there is an essential option of the Voicemail application that is missing. I would like to suggest the implementation of a function to give the user the ability to either allow or disallow the recording of messages. If the ability to record a message is disabled, options u, s, and b must not be considered in order to avoid the playback of messages such as Please leave your message after the tone... the usecase is simple. A person could record a greeting that says please callback later instead of asking to leave a message. usefull also to record afterhour messages. What do you think? Regards, Emrah There is no reason to place this logic in the Voicemail application itself. If you wish to give users the option of leaving a voicemail, it can easily be done in the dialplan by playing prompts and reading the input of the user. Then, based on the input, you can choose whether to run the Voicemail application. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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
Re: [asterisk-users] Setting up Outgoing Trunk
At 3:32 PM on 06 Aug 2009, kumarshantanu wrote: I have a genuine problem in Asterisk setup. I have three inbound trunks in my asterisk box, everything is working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should get different no and if extn: 202 make an outgoing call the recipient should get different one. I'm not sure I understand your question... Are you saying that you want the outbound caller ID to be your public phone number? Or do you want the outbound caller ID to change based on what internal extension initiated the call? If you want your public phone number (the one your telco sends by default) for all outgoing calls, you need to add hidecallerid=yes to your zapata.conf or chan_dahdi.conf for the channels in those trunks. Then Asterisk won't send the caller ID it receives from the internal extensions, and your telco should then insert your phone number as the caller ID on all outbound calls. If you want it set to something other than what your telco puts as the default, you'd have to have hidecallerid=no (or not there) for those channels and set CALLERID(all) in your dialplan for outgoing calls. But this requires that your telco allows you to override the default caller ID. Some do, and some don't--you would have to talk to them. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Set PHP binary location for AGI
Myles Wakeham schrieb: Steve writes: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q This should work. Did you try it? Yes, its working fine. if I can rely on the #! setting in the file, that's good enough for me. It's not really a setting and it has nothing to do with Asterisk. It's just an all normal shebang line. http://en.wikipedia.org/wiki/Shebang_%28Unix%29 http://en.wikipedia.org/wiki/Shebang_(Unix) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] OT - Opensourcesip.org
Anyone have any firsthand experience implementing OpenSBC (opensourcesip.org)? Have a possible consulting gig referral. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com mailto:br...@voipsupply.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers mailto:bsay...@voipsupply.com , CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] No audio on remote SIP calls
Hi, This sounds udp RTP problem. Might be you have some firewall rules that block this kind of traffic ? As soon I remember, Asterisk by default use random port between 1 and 2 for rtp traffic (you can adjust this in rtp.conf). - Sebastien Jonathan Moore escribió: Hi everyone. We have an asterisk server in our main office and phones at each remote site. The remote offices are connected via a MPLS which, to my knowledge has no natting going on. The problem I have is that any call from a remote phone to a remote phone (even on the same remote lan) results in no audio. If I make a call from the same LAN the asterisk server is on, to one of these remote sites, I get perfect two way audio. If I play a call from one phone to another at a remote site, there is no audio, however, I do hear messages (such as voicemail, things from Playback(), etc) that originate on the asterisk server. I've tried adjusting canreinvite= in sip.conf in hopes in might have some effect, but so far nothing. Suggestions on where else to look, or what the problem might be? Which configs would be useful in troubleshooting? Thanks. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 AND NAT
Hello! What are the nat_sip modules you mention? When I set up a linux router some time ago and configured sip.conf with net=yes, everything went smoothly just like any other router. Elliot On Mon, Aug 3, 2009 at 8:45 PM, Gordon Hendersongordon+aster...@drogon.net wrote: On Mon, 3 Aug 2009, Ketema Harris wrote: my questions are: What is the correct way(or resource to find a way) to get a linux firewall to work with SIP so that the NAT issue is not an issue ? Remove all SIP ALG/connection tracking modules and use old fashioned port forwarding on the router and externip=xx.yy.z.qq, localnet= and nat=yes in sip.conf in the asterisk box. That's what I do, anyway. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 1.2 - 1.4 CDR change?
On Thursday 06 August 2009 05:27:57 Chris Bagnall wrote:
Greetings list,
Wondering if anyone can shed some light on an apparent change in CDRs
between 1.2 and 1.4.
One of our clients runs a virtual PA service and has a few hundred DDIs -
one for each client. Their * box is set up with short codes they prefix
their calls with to set the correct outbound CLID, so for example:
123*number_to_be_dialled
This simply does the following:
Set(CALLERID(number)=clid_to_use
Goto(outbound,${EXTEN:4},1)
In * 1.2, this resulted in the callerid field of their CDRs reflecting the
change from their extension number to clid_to_use, which they could then
use to correctly apportion outbound calls to their clients. However, in
1.4, the callerid field retains the original extension number which
initiated the call, which means billing is next to impossible based on
callerid.
While not a change in the callerid management, it is possible that one or more
channels now set CALLERID(ani), in addition to CALLERID(num). The ANI, when
present, is preferred over the number for presentation in the CDR. I
suspect that if you modify this value, the CDR will work as it did before.
--
Tilghman Teryl
with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies)
and Harry, BB, George (dogs)
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Re: [asterisk-users] E1 line simulation for Asterisk
On Thu, Aug 6, 2009 at 7:30 AM, ABBAS SHAKEELshakeel.abbas@gmail.com wrote: thanks alot David that was really helping how can i test for calls etc ie to generate and make calls stuff like that You have four ports. You can use a T1/E1 crossover to send calls from one port to a different port. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] No audio on remote SIP calls
On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien Cramattescrama...@zensoluciones.com wrote: Hi, This sounds udp RTP problem. Might be you have some firewall rules that block this kind of traffic ? As soon I remember, Asterisk by default use random port between 1 and 2 for rtp traffic (you can adjust this in rtp.conf). In theory, there should be no firewalls between my asterisk server and the remote phones. I've opened a ticket with ATT with that exact question, as well as a question of rather any NATing is going on, though, I doubt this is the case, and this is the first time this type of problem has happened in over 4 years. The idea of RTP being to blame would make sense though. I can still transfer and such, and watching the console, I see when I press various keys on the phone, so it seems that the SIP traffic is working out fine. (I do understand that right? SIP == control RTP == voice in a very generic sense?) I plan to take a packet trace in the morning on the asterisk server and see what is going on at that level. Hints as to what I should be looking for? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Monitoring Asterisk uptime
On 7/08/09 2:28 AM, Myles Wakeham wrote: We have added Asterisk to a line of 'mission critical' servers at our business, and being in the web application development business one of the core things we do is to monitor web server availability. I'd like to add Asterisk to the servers that our monitoring systems are handling, and also that our SIP trunk provider has our Asterisk system correctly registered at all times. What are the 'best practice' tricks used for monitoring an Asterisk phone system for uptime and SIP registration from an external monitoring server? I can certainly ping the box, but I really need more than this. I need to know if the Asterisk service is running, and also that there hasn't been any issues with SIP registration to our external trunks. We actually get customers to register to us, and then run a cronjob that checks their registrations are online. If not (they are connected to 5 of our servers) we use clickatel to send SMS messages to our cellphones. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Friday Aug 7th @12 Noon EDT Mobile VoIP
The subject of tomorrow's VoIP Users Conference will be mobile VoIP. If you have any interest, please join us. I myself am tesing a bunch of iPod applications to use with all the usual suspects: OnSIP, Sipgate, Gizmo, Skype, your asterisk box, etc. Details for joining the call are are at http://VUC.me or http://VoipUsersConference.org IRC: #voip-users-conference on Freenode.net See you there. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Anyone had any luck with SIP clients on the iPhone platform?
Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good providers and one has unrecognizable audio and the other has noises as if the cable was badly soldered. I've never experienced such troubles with regular SIP clients. Anyone have any recommendations? Thanks /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 dont detects hangup by phone
Thanks Alot C. Chad Wallace it worked. On Fri, Aug 7, 2009 at 12:09 AM, C. Chad Wallacecwall...@lodgingcompany.com wrote: At 5:24 PM on 06 Aug 2009, ABBAS SHAKEEL wrote: Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this You may need to enable busydetect in chan_dahdi.conf or zapata.conf. However, if you do enable it, I would recommend you also tune it with the busycount and busypattern options so you don't get false positives (asterisk hanging up in the middle of a call). You can read the comments in the sample configs or on voip-info.org for more information: http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf.sample -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Anyone had any luck with SIP clients on the iPhone platform?
Which generation of the handset are you using? They differ in their processing power and that may account for at least some of it. For example, I had a first-generation EDGE iPhone and tried SIP clients over wifi and couldn't get anything useful out of them either. I'm yet to attempt it again with the since-upgraded 3G version but suspect the results might be better. It's hard to say, though. As for VoIP over 3G, well, ATT's 3G + the iPhone's inherent flakyness is so bad down here in LATA 438 as a combination that I don't see that happening, even though I've seen folks pull it off with cheerful success on Sprint and T-Mobile 3G via tether. randulo wrote: Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good providers and one has unrecognizable audio and the other has noises as if the cable was badly soldered. I've never experienced such troubles with regular SIP clients. Anyone have any recommendations? Thanks /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
