[asterisk-users] CDR on MySQL
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten = s-BUSY,n,Playback(useris-curntly-busy)
exten = s-BUSY,n,Hangup(19)
exten = s-CONGESTION,1,NoCDR()
exten = s-CONGESTION,n,Playback(useris-curntly-busy)
exten = s-CONGESTION,n,Hangup(19)
exten = s-CHANUNAVAIL,1,NoCDR()
exten = s-CHANUNAVAIL,n,Playback(useris-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup(19)
exten = s-NOANSWER,1,NoCDR()
exten = s-NOANSWER,n,Playback(number-not-answering)
exten = s-NOANSWER,n,Hangup(19)
;exten = s-ANSWER,1,ResetCDR(w)
;exten = s-ANSWER,n,Set(CDR(UserField)=${SIP_HEADER(From)})
;exten = s-ANSWER,n,Hangup(19)
exten = h,1,DEADAGI(get-unqiueid.php)
on the last line...i would like to get the uniqueid of the call and use
it to compute cost of the call. unfortunately with this setup, after i
hangup, it does not insert the CDR yet. so my AGI get-unqiueid.php does
not find any record. have i placed my ResetCDR(w) correctly?
thank you in advanced.
regards
Ron
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Re: [asterisk-users] CDR on MySQL
On Wed, 2010-12-22 at 18:10 +0800, Ron wrote:
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten = s-BUSY,n,Playback(useris-curntly-busy)
exten = s-BUSY,n,Hangup(19)
exten = s-CONGESTION,1,NoCDR()
exten = s-CONGESTION,n,Playback(useris-curntly-busy)
exten = s-CONGESTION,n,Hangup(19)
exten = s-CHANUNAVAIL,1,NoCDR()
exten = s-CHANUNAVAIL,n,Playback(useris-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup(19)
exten = s-NOANSWER,1,NoCDR()
exten = s-NOANSWER,n,Playback(number-not-answering)
exten = s-NOANSWER,n,Hangup(19)
;exten = s-ANSWER,1,ResetCDR(w)
;exten = s-ANSWER,n,Set(CDR(UserField)=${SIP_HEADER(From)})
;exten = s-ANSWER,n,Hangup(19)
exten = h,1,DEADAGI(get-unqiueid.php)
on the last line...i would like to get the uniqueid of the call and use
it to compute cost of the call. unfortunately with this setup, after i
hangup, it does not insert the CDR yet. so my AGI get-unqiueid.php does
not find any record. have i placed my ResetCDR(w) correctly?
thank you in advanced.
regards
Ron
Make sure you set
endbeforehexten=yes
in cdr.conf
Ish
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[asterisk-users] Maximum E1 Ports on Asterisk ?
Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH -- _ -- 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] Maximum E1 Ports on Asterisk ?
On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo zoelha...@yahoo.co.id wrote: Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it’s not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH Zoel It is possible to do what you are asking. In general the issue is raised about having all your eggs in one basket where one server or hardware failure can drop all of your lines for a period of time. External solutions like Xorcom and Redfone are great ways of abstraction. The concurrent call load on a server relies on the work to be done on each call. If you are using multiple codecs and recording the calls in another file format with other complex dialplan or AGI scripts then one server may not handle the calls well. If everything is ALAW and just dialing though then this would not be a problem for one server. If you search the list for sizing concurrent and load you will find more information. One very nice thing is that testing is very easy with or without the E1 hardware, try running the TDMoE channels between two servers and run a SIPp or other test to see the issues in a lab. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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] Maximum E1 Ports on Asterisk ?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zoel Hairi - Yahoo Sent: Wednesday, December 22, 2010 5:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum E1 Ports on Asterisk ? Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH The general answer is Yes, maybe. I suggest you look at the Xorcom.com website for their load test data. Using a well sized server with best practice tweaks is important. It appears that bigger is not always better. For instance it seems to hurt or at least give no benefit to use a quad core processor. We just ran tests that indicates Xorcom's 3000 model would handle 16 CAS T1s. CAS T1s produce a very high interrupt rate. PRI T1s don't cause nearly as high rate. The right choice of cache, motherboard, processor and tweaks, are essential. You would be leading the pack. I think. .Cary Fitch -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. Please let us know the issue number once raised - I'd like to follow this one. Regards, Steve -- _ -- 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] Asterisk hangs up call after 20s
Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: == WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' == I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying Call established: www.pastebin.com/x6MgnrpG There's also this oddity on line 50: Scheduling destruction of SIP dialog. FWIW, in sip.conf, for the remote XLite user, I tried nat=no and nat=yes, with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using t1min=500 in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated. -- _ -- 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] Maximum E1 Ports on Asterisk ?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, December 22, 2010 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ? On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo zoelha...@yahoo.co.id wrote: Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH Zoel It is possible to do what you are asking. In general the issue is raised about having all your eggs in one basket where one server or hardware failure can drop all of your lines for a period of time. External solutions like Xorcom and Redfone are great ways of abstraction. The concurrent call load on a server relies on the work to be done on each call. If you are using multiple codecs and recording the calls in another file format with other complex dialplan or AGI scripts then one server may not handle the calls well. If everything is ALAW and just dialing though then this would not be a problem for one server. If you search the list for sizing concurrent and load you will find more information. One very nice thing is that testing is very easy with or without the E1 hardware, try running the TDMoE channels between two servers and run a SIPp or other test to see the issues in a lab. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ In a previous post I also mentioned Xorcom. They do have a unique fail over ability with their Astribank systems. With dual servers, separate chassis and power supplies for the 4 port T1/E1 cards, USB interconnections, and redundant power supplies for the Astribanks, system downtime can be minimized, and if there is a failure, repair would be at worst, no screwdriver needed. If system failure would be idling 200 - 400 people, avoiding system down time would be a major objective. Cary Fitch -- _ -- 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] Asterisk hangs up call after 20s
On 22 December 2010 12:44, Gilles codecompl...@free.fr wrote: Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: == WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' == I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying Call established: www.pastebin.com/x6MgnrpG There's also this oddity on line 50: Scheduling destruction of SIP dialog. FWIW, in sip.conf, for the remote XLite user, I tried nat=no and nat=yes, with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using t1min=500 in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated. Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Cheers, Steve -- _ -- 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] Asterisk hangs up call after 20s
Hello, you have a typicall nat issue. Asterisk receives messages from the phone but cannot send any messages back (thats why it tries to resend the 200 ok message 6 times). try setting qualify=yes to your sip peers config to keep the nat port open. best regards stefan Am 22.12.10 13:44, schrieb Gilles: Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: == WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' == I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying Call established: www.pastebin.com/x6MgnrpG There's also this oddity on line 50: Scheduling destruction of SIP dialog. FWIW, in sip.conf, for the remote XLite user, I tried nat=no and nat=yes, with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using t1min=500 in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated. -- _ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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] Simplifying dial-plan
On Wed, Dec 22, 2010 at 2:01 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote:
Hi Stephen,
_NXXNXX
_NXX
_011.
_911
Of course it can, but it depends on what you want to do when those
numbers are called...
I didn't know about the setvar in the sip.conf and actually I think it
is a much cleaner solution. Since you are already using it I would
suggest to not only use it for CallerID but also for the
@vitel-outbound like this:
exten = _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@${outbound})
exten = _1NXXNXX,n,Goto(h,1)
Of course you'll need to set setvar=Outbound=vitel-outbound or
setvar=Outbound=vitel-outbound2 in sip.conf.
What do you want to do with the other numbers? If you want to do the
same as with _1NXXNXX you can just add things like this in your
extensions.conf:
exten = _NXXNXX,1,Goto(_1NXXNXX,1)
exten = 911,1,Goto(_1NXXNXX,1)
Or you can do different things if you want that like this:
exten = _NXX,1,Set(CALLERID(all)=No one cares 0)
exten = _NXX,n,Dial(SIP/${ext...@abcdefgh)
exten = _NXX,n,Goto(h,1)
Best regards,
Jeroen Eeuwes
Jeroen, I'm trying to avoid rewriting the outgoing block for the
patterns mentioned above. I've placed a pseudo dial-plan below. The
plan needs to dial the 1 and/or also the area code depending on the
pattern they enter. Any tips, thanks.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _NXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _NXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _NXXNXX,n,Goto(h,1)
exten = _NXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
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[asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London
Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: 1. Candidate must have good knowledge of setting up SIP and IAX Trunks. 2. Must have experience in installing and configuring SIP Express Router or OPEN SER. 3. Installation and trouble shooting of Asterisk Servers using Centos. 4. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. 5. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. 6. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. 7. Scripting / Bash scripting would be useful. 8. Expert knowledge in Configuring, Maintaining and querying MySQL. 9. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.netmailto:ja...@langleyjames.co.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] Include ${HANGUPCAUSE} in CDR
I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a
field called cause_code but it won't write. I belive it is because the
record has already been written by the time I hit the h section of the
code. How might I get this info into the CDR. I need this info for Quality
of Service as well as route checking. Any ideas would be apperciated.
Here is my dial line and my h lines. I also use the g option so if the
other party hangs up and that is not working either.
exten =
doDialStd,n,Dial(${siteDefaultOutboundTrunk}/${c_DialArg}${c_DialExten},120,
ge)
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
Bryant exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})Bryant
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Re: [asterisk-users] DIALSTATUS on CANCEL
Hi Nikhil, Both debug and verbose are set to 20. That's all I got, but as you can see, for the other types of reasons, the DIALSTATUS got a value (and we see the events). I'm pretty sure it's a bug. Michael On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net wrote: Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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] CDR on MySQL
Hi I have tried setting endbeforehexten=yes but still CDR does not get
inserted before h exten. what i tried is setting ResetCDR(w) before the
DEADAGI. Like this:
exten = h,1,ResetCDR(w)
exten = h,2,DEADAGI(get-unqiueid.php)
it seems to work but it's inserting 2 record on the CDR, one with
disposition ANSWERED and one with NO ANSWER.
any ideas? thanks again.
regards
Ron
On 12/22/2010 7:29 PM, Ishfaq Malik wrote:
On Wed, 2010-12-22 at 18:10 +0800, Ron wrote:
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten = s-BUSY,n,Playback(useris-curntly-busy)
exten = s-BUSY,n,Hangup(19)
exten = s-CONGESTION,1,NoCDR()
exten = s-CONGESTION,n,Playback(useris-curntly-busy)
exten = s-CONGESTION,n,Hangup(19)
exten = s-CHANUNAVAIL,1,NoCDR()
exten = s-CHANUNAVAIL,n,Playback(useris-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup(19)
exten = s-NOANSWER,1,NoCDR()
exten = s-NOANSWER,n,Playback(number-not-answering)
exten = s-NOANSWER,n,Hangup(19)
;exten = s-ANSWER,1,ResetCDR(w)
;exten = s-ANSWER,n,Set(CDR(UserField)=${SIP_HEADER(From)})
;exten = s-ANSWER,n,Hangup(19)
exten = h,1,DEADAGI(get-unqiueid.php)
on the last line...i would like to get the uniqueid of the call and use
it to compute cost of the call. unfortunately with this setup, after i
hangup, it does not insert the CDR yet. so my AGI get-unqiueid.php does
not find any record. have i placed my ResetCDR(w) correctly?
thank you in advanced.
regards
Ron
Make sure you set
endbeforehexten=yes
in cdr.conf
Ish
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[asterisk-users] dahdi-channels.conf for Digium TDM2400
Hello everyone, I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line=2 WCTDM/0/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by DAHDI tools, and since it's working I didn't want to change it. Is it safe to remove them? Thanks in advance, Alex Saavedra -- _ -- 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] callerid and user on voicemail
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Wednesday, December 22, 2010 4:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] callerid and user on voicemail
Hello,
There is a problem that i can not figure out how to solve.
I got users with 5 digit usernames for sip.
Some users has a callerid for outside calls.
I have such problems
When a user activates (for ex) call forwarding, System creates that entry on
database as CFIM/callerid not the username,
So this rule works only if a call is made from outside to the callerid. Not
the local calls made to username.
Or, if that user dials *97 and tries to enter voicemail, voicemail
application looks for callerid instead of username , so it can not find it.
And got similar problems in some other applications too.
So, how can i make to use callerid only for outbound calls, or to forward
incoming calls to local extensions.
This won't completely solve your questions, but here are some tips. #1.
You can define a different callerid than the user-id in sip.conf. For
example, your user 12345 may look like this
[12345]
Type=peer
Context=default
Add this line
Callerid=Joe Cool 5551212
#2. *97 is just a dialplan line like this:
Exten = *97,1,voicemailmain(${CALLERID(n...@default)
You can either do some error trapping or use ex-girlfriend logic like this
Exten = *97,1,noop(new *97 logic)
Exten = *97/12345,n,voicemailmain(1...@default)
Exten = *97,n,voicemailmain(${CALLERID(num)@default)
Hope this is useful.
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[asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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] setting up callerid
Hi Dave, context=openbts callerid=473520 I see you are using OpenBTS. To my understanding, OpenBTS does not support caller ID, so I don't think it can work. But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know. Disregard my answer. I just tested the callerid on my OpenBTS and it worked. So the problem you encounter must be elsewhere. Regards Axelle -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
I'm going to guess you aren't going to get a lot of help on a list hosted by Digium on how to use a potentially illegal codec... That said, ast14 in the filename might signify what the problem is. The APIs likely changed for modules between 1.4 and 1.8. On Wed, Dec 22, 2010 at 7:58 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
I don't think a module compiled for Asterisk 1.4 will work with any other Asterisk version. On Wed, Dec 22, 2010 at 8:58 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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] DIALSTATUS on CANCEL
I see the same thing. Why is there an CANCEL status if it is never set. The only way I have been able to capture a Cancel status is with the h extensions using the 'e' option under dial. But this leaves no way to tell what the DIALSTATUS state was as it is blank. I belive it is a bug as well. Bryant From: Michael voip.quest...@gmail.com Sent: Wednesday, December 22, 2010 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS on CANCEL Hi Nikhil, Both debug and verbose are set to 20. That's all I got, but as you can see, for the other types of reasons, the DIALSTATUS got a value (and we see the events). I'm pretty sure it's a bug. Michael On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net wrote: Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
To my knowledge there is currently no free version of the g729 codec. There were some spec builds but those were just for testing if I recall correctly. Each version of the codec that we have always gotten has been compiled for each version of asterisk. I would just buy the Digium licenses for the codec and not mess with it. That way you are legal and have support if you need it. From: Joel Maslak jmas...@antelope.net Sent: Wednesday, December 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!) I'm going to guess you aren't going to get a lot of help on a list hosted by Digium on how to use a potentially illegal codec... That said, ast14 in the filename might signify what the problem is. The APIs likely changed for modules between 1.4 and 1.8. On Wed, Dec 22, 2010 at 7:58 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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 -- _ -- 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 list used extensions + assign extension to a roaming phone
Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do dialplan show that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' = 1. Macro(dialSIP|IMSI1) [pbx_config] '2103' = 1. Macro(dialSIP|IMSI2) [pbx_config] '2104' = 1. Macro(dialSIP|IMSI3) [pbx_config] but it does not tell me who is actually registered or using the network, maybe only 2102. - is it possible to assign a given number/range of numbers (extension) to a phone which roams into my network (open registration)? Thanks Axelle -- _ -- 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] DIALSTATUS on CANCEL
Try to use h extension -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Michael wrote: Hi Nikhil, Both debug and verbose are set to 20. That's all I got, but as you can see, for the other types of reasons, the DIALSTATUS got a value (and we see the events). I'm pretty sure it's a bug. Michael On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
Hi all, thanks for answering. You all are right but I do not really need the codec because my phones and my Voip lines are all working using g729. Asterisk is working fine without transcoding as well.the problem is my CLI is flooded with messages like: WARNING[7831] translate.c: No translator path from alaw to unknown which are quite annoying...aren't they? Should I pay to avoid a CLI message? That doesn't sound fair to me. I know I should report the problem but the fake codec seemed the faster way. Giorgio Incantalupo Giorgio Incantalupo wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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] How to list used extensions + assign extension toa roaming phone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Wednesday, December 22, 2010 9:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to list used extensions + assign extension toa
roaming phone
Hi list,
I have searched through asterisk command lines but haven't found how to do
this:
- can I list the phones (callerid or IMSIs?) currently registered ?
If I do dialplan show that lists the configuration I loaded, e.g
[ Context 'sip-local' created by 'pbx_config' ]
'2102' = 1. Macro(dialSIP|IMSI1) [pbx_config]
'2103' = 1. Macro(dialSIP|IMSI2) [pbx_config]
'2104' = 1. Macro(dialSIP|IMSI3) [pbx_config]
but it does not tell me who is actually registered or using the
network, maybe only 2102.
- is it possible to assign a given number/range of numbers (extension)
to a phone which roams into my network (open registration)?
Thanks
Axelle
For question 1, I think sip show peers is what you want. For question 2,
here are two ways to do it.
#1 dial fixed number using 2000-2999
Exten = 2xxx,1,dial(SIP/foo)
#2 assign number using 3001, then dial it with 2000-2999
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,assignroam,3)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Set(ROAM=${digito})
exten = 3001,n,Set(DB(roam/ext)=${digito})
exten = 3001,n,playback(vm-goodbye)
exten = 3001,n,hangup
exten = 2xxx,1,Set(ROAM=${DB(roam/ext)})
exten = 2xxx,n,dial(SIP/${ROAM})
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Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
Hi MrHanMan, you are right...and the mistake is so stupid I've already solved itwhat a slip! :) This means I really need a long relaxing period on some exotic island...or in some cold prison since I'm using an illegal codec!!! :) Still I do not believe why Asterisk had not complained for a different version module instead of asking for a license. Should I report this weirdness? Btw thank you for your time. Giorgio Incantalupo P.S.: as I've already written in some other post, I use the criminal codec to test Voip lines without the need to install the license every time and to avoid a noisy message flooding my CLI. MrHanMan wrote: I don't think a module compiled for Asterisk 1.4 will work with any other Asterisk version. On Wed, Dec 22, 2010 at 8:58 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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 -- _ -- 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] DIALSTATUS on CANCEL
The Dial Status is not set when accessing it from the h extension. Bryant From: Vardan Harutyunyan hvarda...@gmail.com Sent: Wednesday, December 22, 2010 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS on CANCEL Try to use h extension -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Michael wrote: Hi Nikhil, Both debug and verbose are set to 20. That's all I got, but as you can see, for the other types of reasons, the DIALSTATUS got a value (and we see the events). I'm pretty sure it's a bug. Michael On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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 -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Wednesday, December 22, 2010 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!) Hi all, thanks for answering. You all are right but I do not really need the codec because my phones and my Voip lines are all working using g729. Asterisk is working fine without transcoding as well.the problem is my CLI is flooded with messages like: WARNING[7831] translate.c: No translator path from alaw to unknown which are quite annoying...aren't they? Should I pay to avoid a CLI message? That doesn't sound fair to me. I know I should report the problem but the fake codec seemed the faster way. Giorgio Incantalupo snip Why don't you just modify that piece of code to kill the warning? Are you using a canned Asterisk (AsteriskNow/FreePBX/etc)? -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer 45K South London
45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: 1.Candidate must have good knowledge of setting up SIP and IAX Trunks. 2.Must have experience in installing and configuring SIP Express Router or OPEN SER. 3.Installation and trouble shooting of Asterisk Servers using Centos. 4.Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. 5.Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. 6.Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. 7.Scripting /Bash scripting would be useful. 8.Expert knowledge in Configuring, Maintaining andqueryingMySQL. 9.Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net [mailto:ja...@langleyjames.co.uk] Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
Giorgio You could buy just a couple of licenses 3 to 5. It would get rid of the messages for the most part and it would give you the ability to transcode for voicemails and other items requiring transcode. The reason you are likely getting the messages is there is some kind of transcode required that it can't do and you are getting the warring. If you shut off all in the middle functions like recording, voicemail, and feature codes you may be able to get rid of them but you would also loose the functions. You will likely waste more than the $30 to $50 dollars in time and you get the option to transcode to boot. Just my 2 cents. From: Giorgio Incantalupo gincantal...@fgasoftware.com Sent: Wednesday, December 22, 2010 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!) Hi all, thanks for answering. You all are right but I do not really need the codec because my phones and my Voip lines are all working using g729. Asterisk is working fine without transcoding as well.the problem is my CLI is flooded with messages like: WARNING[7831] translate.c: No translator path from alaw to unknown which are quite annoying...aren't they? Should I pay to avoid a CLI message? That doesn't sound fair to me. I know I should report the problem but the fake codec seemed the faster way. Giorgio Incantalupo Giorgio Incantalupo wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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 -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
On Wed, Dec 22, 2010 at 9:48 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: Hi all, thanks for answering. You all are right but I do not really need the codec because my phones and my Voip lines are all working using g729. Asterisk is working fine without transcoding as well.the problem is my CLI is flooded with messages like: WARNING[7831] translate.c: No translator path from alaw to unknown which are quite annoying...aren't they? Should I pay to avoid a CLI message? That doesn't sound fair to me. I know I should report the problem but the fake codec seemed the faster way. Hmm...potentially infringing on a patent or adding a noload directive to modules.conf. It's a toss up! ;) -Michael -- _ -- 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] dahdi-channels.conf for Digium TDM2400
Hi, On Wed, Dec 22, 2010 at 9:49 AM, Alex Saavedra a...@masterline-logistics.com wrote: I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line=2 WCTDM/0/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by DAHDI tools, and since it's working I didn't want to change it. Is it safe to remove them? Short Answer: NO!! Longer Answer: The settings all apply to channels, which are defined by the channel = 2 directive. If I'm remembering correctly, the channel is set at the end of the Stanza, not at the beginning. So, your blank callerid and group would apply to your next channel directive (3?). Now, I remember reading there is a way to flip the channel definition bit (channel = XX) to the top of the stanza, but can't recall. Now, if in between two channel definitions you have repetition, it might be ok to trim things up, as long as it has the right information -- the last setting is the effective one. And the bit that starts ;;; is a comment, which is actually ignored by asterisk. Hope this helps, Gerald. -- _ -- 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] Simplifying dial-plan
Hi Stephen,
Jeroen, I'm trying to avoid rewriting the outgoing block for the
patterns mentioned above. I've placed a pseudo dial-plan below. The
plan needs to dial the 1 and/or also the area code depending on the
pattern they enter. Any tips, thanks.
I find a diaplan much easier to read if all the lines of the same
exten = pattern,x,etc are grouped. The way you showed it now is
very difficult to read.
What you could do is something like this
exten = _NXXNXX,1,Goto(1${EXTEN},1)
exten = _NXX,1,Goto(1555${EXTEN},1)
Assuming that you want to dial an 1 if not dialed first and you've got
10 digits or if you only receive 7 digits you want to add both an 1
and areacode 555.
Because you've added the extra digits yourself it will match to the
_1NXXNXX extension and start there at 1.
Best regards,
Jeroen Eeuwes
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Re: [asterisk-users] Asterisk hangs up call after 20s
On Wed, 22 Dec 2010 14:31:32 +0100, Stefan Schmidt s...@sil.at wrote: you have a typicall nat issue. Asterisk receives messages from the phone but cannot send any messages back (thats why it tries to resend the 200 ok message 6 times). try setting qualify=yes to your sip peers config to keep the nat port open. Thanks for the idea, but all users are defined with qualify=yes: = /etc/asterisk cat sip.conf [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = yes ;allowexternalinvites=yes externip=public IP localnet=192.168.0.0/24 ;Other IPs can still REGISTER :-/ deny=0.0.0.0/0 permit=VOSP IP/255.255.255.255 permit = 192.168.0.0/255.255.255.0 alwaysauthreject=yes ;for safety context = dummmy ;all RTP packets go through Asterisk canreinvite=no ;makes no difference: still hangs up ;t1min=500 disallow=all allow=ulaw allow=alaw allow=gsm register = me:p...@vosp.com [vosp_outgoing] type=peer host=vosp.com username=me secret=mysecret fromuser=me fromdomain=vosp.com nat=yes canreinvite=no qualify=yes [vosp_incoming] type=peer host=vosp.com context=from_vosp nat=yes canreinvite=no insecure=port,invite qualify=yes ;(!) means it's a template [sets](!) type=friend context=my-phones host=dynamic qualify=yes nat=no [local-xlite](sets) secret=mysecret [remote-xlite](sets) secret=mysecret ;remote extension behind own NAT: nat=yes or nat=no? ;makes no difference : still hangs up ;nat=yes nat=no = What's weird, is that the remote XLite can successfully call the local XLite and I get sound both ways, and it's only 20s into the call that Asterisk decides to give up and hang up (while the remote side still thinks everything's OK). I tried SJphone instead of XLite, same result. Could it be some wrong configuration in Asterisk? Thank you. -- _ -- 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] Asterisk hangs up call after 20s
On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com wrote: Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Thanks for the tip, but I get the same problem with SJPhone and PhonerLite, so it looks like a problem in Asterisk. -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some values
to the CDR then.
For the write of the CDR and stopping the billing seconds. Then continue
to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant I need the cause code stored.Really what I need to be able to do is
in the h quickly store some values to the CDR then.For the write of the CDR
and stopping the billing seconds. Then continue to process some cleanup
funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
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Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
On Wednesday 22 December 2010 08:23:19 MrHanMan wrote: On Wed, Dec 22, 2010 at 6:41 AM, Steve Davies davies...@gmail.com wrote: On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. Please let us know the issue number once raised - I'd like to follow this one. I happened to see it pop up on the bug tracker. Issue #0018515. Very funny error message in the patch. It's a forward-port of a section of code that was in res_agi in 1.4. It was no longer needed in res_agi because AGIs can now continue to interact with Asterisk after a hangup event, transitioning gracefully into DeadAGI. -- Tilghman -- _ -- 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] CDR on MySQL
What would it do if you
exten = h,1,ResetCDR(w)
exten = h,2,NoCDR()
exten = h,3,DEADAGI(get-unqiueid.php)
I have not tried it but in theory it should write the first CDR and then
kill the write of the second NO ANSWER CDR.
Let me know if it works for you as I may need to do it on some of my h
exten code as well.
Bryant
From: Ron nha...@gmail.com
Sent: Wednesday, December 22, 2010 9:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CDR on MySQL
Hi I have tried setting endbeforehexten=yes but still CDR does not get
inserted before h exten. what i tried is setting ResetCDR(w) before the
DEADAGI. Like this:
exten = h,1,ResetCDR(w)
exten = h,2,DEADAGI(get-unqiueid.php)
it seems to work but it's inserting 2 record on the CDR, one with
disposition ANSWERED and one with NO ANSWER.
any ideas? thanks again.
regards
Ron
On 12/22/2010 7:29 PM, Ishfaq Malik wrote:
On Wed, 2010-12-22 at 18:10 +0800, Ron wrote:
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten = s-BUSY,n,Playback(useris-curntly-busy)
exten = s-BUSY,n,Hangup(19)
exten = s-CONGESTION,1,NoCDR()
exten = s-CONGESTION,n,Playback(useris-curntly-busy)
exten = s-CONGESTION,n,Hangup(19)
exten = s-CHANUNAVAIL,1,NoCDR()
exten = s-CHANUNAVAIL,n,Playback(useris-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup(19)
exten = s-NOANSWER,1,NoCDR()
exten = s-NOANSWER,n,Playback(number-not-answering)
exten = s-NOANSWER,n,Hangup(19)
;exten = s-ANSWER,1,ResetCDR(w)
;exten = s-ANSWER,n,Set(CDR(UserField)=${SIP_HEADER(From)})
;exten = s-ANSWER,n,Hangup(19)
exten = h,1,DEADAGI(get-unqiueid.php)
on the last line...i would like to get the uniqueid of the call and use
it to compute cost of the call. unfortunately with this setup, after i
hangup, it does not insert the CDR yet. so my AGI get-unqiueid.php does
not find any record. have i placed my ResetCDR(w) correctly?
thank you in advanced.
regards
Ron
Make sure you set
endbeforehexten=yes
in cdr.conf
Ish
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Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London
On Wed, 2010-12-22 at 11:23 -0500, C. Savinovich wrote: 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS And you have to know Kalamino! :) j On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: 1. Candidate must have good knowledge of setting up SIP and IAX Trunks. 2. Must have experience in installing and configuring SIP Express Router or OPEN SER. 3. Installation and trouble shooting of Asterisk Servers using Centos. 4. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. 5. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. 6. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. 7. Scripting / Bash scripting would be useful. 8. Expert knowledge in Configuring, Maintaining and querying MySQL. 9. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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] Simplifying dial-plan
On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese rsre...@gmail.com wrote:
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
Sorry, here's the rest.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
Why not make a Macro (or GoSub) to handle this block of code, and then your
outbound dial lines are just one line calling the Macro? Saves a lot of
repeating blocks of code. Something like this (not tested):
[macro-OutboundDial]
; ${ARG1} = CHANNEL
; ${ARG2} = EXTERNAL_CALLERID
exten = s,1,Set(Outgoing=${CUT(${ARG1},/,2)})
exten = s,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = s,n,GotoIf($[${Outgoing} = 201]?outbound2:outbound1)
exten = s,n(outbound1),Set(CALLERID(all)=${ARG2})
exten = s,n,Dial(SIP/${macro_ext...@vitel-outbound)
exten = s,n,Goto(h,1)
exten = s,n(outbound2),Set(CALLERID(all)=${ARG2})
exten = s,n,Dial(SIP/${macro_ext...@vitel-outbound2)
exten = s,n,Goto(h,1)
[outbound-context]
exten = _NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _NXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
2010/12/22 Bryant Zimmerman brya...@zktech.com Giorgio You could buy just a couple of licenses 3 to 5. It would get rid of the messages for the most part and it would give you the ability to transcode for voicemails and other items requiring transcode. The reason you are likely getting the messages is there is some kind of transcode required that it can't do and you are getting the warring. If you shut off all in the middle functions like recording, voicemail, and feature codes you may be able to get rid of them but you would also loose the functions. You will likely waste more than the $30 to $50 dollars in time and you get the option to transcode to boot. Just my 2 cents. Mayba I'm hijacking this thread, but what about virtual machines ? At the moment, let say you're using an hardware platform on which you launch virtual machines (one per project but only one at a time). Would a single licence be usable on each virtual machine (same (virtualized?) processor and mac addresses) ? -- _ -- 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] Siemens OpenStage phones and Asterisk
Hi, Any recent experience to share when using OpenStage phones in SIP mode and Asterisk ? What about provisionning (and localization) ? BLF ? Audio quality ? User acceptance ? Cheers -- _ -- 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] Asterisk hangs up call after 20s
This is a NAT issue like noted before. Try: localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0 instead of: localnet=192.168.0.0/24 http://192.168.0.0/24Also, make sure you have all your VPN connections as localnet and other side subnet as localnet as well if you are using VPN. Otherwise, open the neccessary ports needed for SIP and RTP. If you note your router type someone might be able to help more specifically. -Bruce On Wed, Dec 22, 2010 at 12:27 PM, Gilles codecompl...@free.fr wrote: On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com wrote: Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Thanks for the tip, but I get the same problem with SJPhone and PhonerLite, so it looks like a problem in Asterisk. -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer 45K South London
45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Wednesday, December 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some
values to the CDR then.
For the write of the CDR and stopping the billing seconds. Then
continue to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in
the h quickly store some values to the CDR then.For the write of the
CDR and stopping the billing seconds. Then continue to process some
cleanup funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Is the CDR line your only h line? I ask because if you only have one
priority for h then you MUST have:
exten = h,1,Set(CDR(cause_code)=${HANGUPCAUSE})
This is because the dialplan will not use n for the first priority and
thus will never run.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, December 22, 2010 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Developers Mailing List
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are finished
; executing. By enabling this option, the CDR will be ended before executing
; the h extension so that CDR values such as end and billsec may be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some values to
the CDR then.
For the write of the CDR and stopping the billing seconds. Then continue to
process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in the h
quickly store some values to the CDR then.For the write of the CDR and
stopping the billing seconds. Then continue to process some cleanup
funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Okay, this is a mentally challenged solution, but at the h extension you
have ${UNIQUEID} and ${HANGUPCAUSE} available to you. Use DeadAGI to
stuff these values into the CDR or into another file that you can
cross-reference. If you use the two-file method, you would just select the
matching file using uniqueid. That's what I did when I had this problem on
a version of 1.4.
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Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, December 22, 2010 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London On Wed, 2010-12-22 at 11:23 -0500, C. Savinovich wrote: 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS And you have to know Kalamino! :) You know Kalamino? I haven't seen him since prison in Budapest! Tell him hello! :-) CF -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be ; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Sounds like your h extension is in the wrong context. Try including some
information about where you are putting the h extension and what includes
you're doing.
--
Tilghman
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Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
My understanding is that you need one license for every channel it's being used on, regardless of whether the server is physical or virtual. On Wed, Dec 22, 2010 at 12:20 PM, Olivier oza_4...@yahoo.fr wrote: 2010/12/22 Bryant Zimmerman brya...@zktech.com Giorgio You could buy just a couple of licenses 3 to 5. It would get rid of the messages for the most part and it would give you the ability to transcode for voicemails and other items requiring transcode. The reason you are likely getting the messages is there is some kind of transcode required that it can't do and you are getting the warring. If you shut off all in the middle functions like recording, voicemail, and feature codes you may be able to get rid of them but you would also loose the functions. You will likely waste more than the $30 to $50 dollars in time and you get the option to transcode to boot. Just my 2 cents. Mayba I'm hijacking this thread, but what about virtual machines ? At the moment, let say you're using an hardware platform on which you launch virtual machines (one per project but only one at a time). Would a single licence be usable on each virtual machine (same (virtualized?) processor and mac addresses) ? -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
On Wednesday 22 December 2010 12:20:36 Olivier wrote: 2010/12/22 Bryant Zimmerman brya...@zktech.com Giorgio You could buy just a couple of licenses 3 to 5. It would get rid of the messages for the most part and it would give you the ability to transcode for voicemails and other items requiring transcode. The reason you are likely getting the messages is there is some kind of transcode required that it can't do and you are getting the warring. If you shut off all in the middle functions like recording, voicemail, and feature codes you may be able to get rid of them but you would also loose the functions. You will likely waste more than the $30 to $50 dollars in time and you get the option to transcode to boot. Just my 2 cents. Mayba I'm hijacking this thread, but what about virtual machines ? At the moment, let say you're using an hardware platform on which you launch virtual machines (one per project but only one at a time). Would a single licence be usable on each virtual machine (same (virtualized?) processor and mac addresses) ? No, because each virtual machine gets its own virtualized Ethernet MAC address. -- Tilghman -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer 45K South London
C. Savinovich wrote: 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS Isn't that in UK money? Or Euros? JN On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: *Job Description: Asterisk MySQL Support Engineer* Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: 1. Candidate must have good knowledge of setting up SIP and IAX Trunks. 2. Must have experience in installing and configuring SIP Express Router or OPEN SER. 3. Installation and trouble shooting of Asterisk Servers using Centos. 4. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. 5. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. 6. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. 7. Scripting / Bash scripting would be useful. 8. Expert knowledge in Configuring, Maintaining and querying MySQL. 9. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 * * *Jess Hart * __* * *Langley James IT Recruitment* 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer 45KSouth London
Wouldn't that be 70K USD? Or should we REALLY be worried about the British economy? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Wednesday, December 22, 2010 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Wednesday, December 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer 45K South London
Can you point out to me the places in London that sell food at American prices? Perhaps I get SeamlessWeb to deliver every morning from Brooklyn to London. On December 22, 2010 at 1:24 PM Don Kelly d...@donkelly.biz wrote: 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfC. Savinovich Sent:Wednesday, December 22, 2010 10:23 AM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net [mailto:ja...@langleyjames.co.uk] Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- 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] Siemens OpenStage phones and Asterisk
Hi! Any recent experience to share when using OpenStage phones in SIP mode and Asterisk ? I found these phones not to be very comfortable to use, even though they do look interesting and the hardware is well done. If I remember well you can run your on JAVA (and/or XML) applications on it - loading it the first time takes a while, though. What about provisionning (and localization) ? Provisioning is certainly a strong feature of those phones, but it is also complex. Look at the Asterisk solution Gemeinschaft if you are interested in a PBX product that has attacked this. BLF ? Working but limited (during my short test I think I didn't get pickup-with-BLF-button working). It is very well possible that with a newer firmware this was addressed. See also: http://wiki.siemens- enterprise.com/index.php/Asterisk_Feature_Busy_Lamp_Field_%28BLF%29 Audio quality ? Good or very good. See: http://wiki.siemens-enterprise.com/index.php/OpenStage_and_Asterisk http://www.voip-info.org/wiki/view/Siemens+Phones P.S.: Last time I checked the firmware was not easily available from Siemens. -- _ -- 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] dahdi-channels.conf for Digium TDM2400
Gerald, Thank you for the explanation. Glad I asked... Alex Saavedra On Wed, Dec 22, 2010 at 12:40 PM, Gerald A geraldabli...@gmail.com wrote: Hi, On Wed, Dec 22, 2010 at 9:49 AM, Alex Saavedra a...@masterline-logistics.com wrote: I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line=2 WCTDM/0/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by DAHDI tools, and since it's working I didn't want to change it. Is it safe to remove them? Short Answer: NO!! Longer Answer: The settings all apply to channels, which are defined by the channel = 2 directive. If I'm remembering correctly, the channel is set at the end of the Stanza, not at the beginning. So, your blank callerid and group would apply to your next channel directive (3?). Now, I remember reading there is a way to flip the channel definition bit (channel = XX) to the top of the stanza, but can't recall. Now, if in between two channel definitions you have repetition, it might be ok to trim things up, as long as it has the right information -- the last setting is the effective one. And the bit that starts ;;; is a comment, which is actually ignored by asterisk. Hope this helps, Gerald. -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer45KSouth London
Wait, is 70k US for an experienced engineer supposed to be adequate? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, December 22, 2010 2:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London Wouldn't that be 70K USD? Or should we REALLY be worried about the British economy? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Wednesday, December 22, 2010 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Wednesday, December 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com --
[asterisk-users] Asterisk 1.8.1.1 Multiple Parking Lots
Asterisk Version: 1.8.1.1 Problem: Multiple Parking Lots Issue: Not redirecting to the right parking lot. Always uses the first parking lot from parkedcalls show or features show Asterisk Working Version: 1.6.1 Steps Taken: In features.conf added: [parkinglot_test] context = parkedcalls-test parkext = 700 parkpos = 701-710 parkingtime = 120 findslot = next In extensions.include at the bottom of [local-extensions-test]: exten = 701,hint,park:7...@parkedcalls-test exten = 702,hint,park:7...@parkedcalls-test In extensions.include in [from-inside-redir-test] and [from-inside-restricted-redir-test]: include = parkedcalls-test For each extension in sip.conf in [XXX-test]: In between the existing 'vmexten' and 'secret' lines parkinglot=parkinglot_test Output from files sip show peer 101-test.txt and sip show peer 102-test.txt shows that parking lot is set to parkinglot_test sip show peer 101-test CODE: SELECT ALL voice2*CLI sip show peer 101-test * Name : 101-test Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-inside-test Subscr.Cont. : local-extensions-test Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1...@default-test VM Extension : 101 LastMsgsSent : 32767/65535 Call limit : 99 Max forwards : 0 Dynamic : Yes Callerid : Test Tenant 101 MaxCallBR: 384 kbps Expire : 956 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.211.0.42:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 101-test SIP Options : 100rel gruu path replaces replace timer Codecs : 0x106 (gsm|ulaw|g729) Codec Order : (g729:20,ulaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (34 ms) Useragent: Aastra 55i/2.6.0.1008 Reg. Contact : sip:101-t...@10.211.0.42:5060;transport=udp Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : parkinglot_test Use Reason : No Encryption : No sip show peer 102-test CODE: SELECT ALL voice2*CLI sip show peer 102-test * Name : 102-test Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-inside-test Subscr.Cont. : local-extensions-test Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1...@default-test VM Extension : 102 LastMsgsSent : 32767/65535 Call limit : 99 Max forwards : 0 Dynamic : Yes Callerid : Test Tenant 102 MaxCallBR: 384 kbps Expire : 2363 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.211.0.41:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 102-test SIP Options : (none) Codecs : 0x106 (gsm|ulaw|g729) Codec Order : (g729:20,ulaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (31 ms) Useragent: Aastra 55i/2.6.0.1008 Reg. Contact : sip:102-t...@10.211.0.41:5060;transport=udp Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : parkinglot_test Use Reason : No Encryption : No File debug.txt shows that when a call is parked, it is NOT sending over to parkinglot_test, but parkinglot_fitts. Debug.txt CODE: SELECT ALL == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Called 101-test == Extension Changed 101[local-extensions-test] new state Ringing for Notify User 102-test -- SIP/101-test-0035 is ringing -- SIP/101-test-0035 is ringing == Extension Changed 101[local-extensions-test] new state InUse for Notify User 102-test -- SIP/101-test-0035 answered SIP/DASH_SIP_TRUNK_DENVER-0033 -- Started music on hold, class 'default-test', on SIP/DASH_SIP_TRUNK_DENVER-0033 == Extension Changed
Re: [asterisk-users] Vacancy - Asterisk MySQL SupportEngineer45KSouth London
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Watkins, Bradley Sent: Wednesday, December 22, 2010 2:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL SupportEngineer45KSouth London Wait, is 70k US for an experienced engineer supposed to be adequate? Offhand, I'd jump on it, but in a city like London, it probably wouldn't keep you off the street. -- _ -- 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] Echo Cancellation Problem - Invalid Argument?!?
Well, I downgraded this box to Asterisk 1.4.38 and all is well again. Echo cancellation works properly, no problems, no errors. I have to assume this is a bug in Asterisk 1.8.x or Wanpipe 3.5.18. --Tim - Original Message - Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 9 (Invalid argument) Relevant components: Asterisk: Asterisk SVN-trunk-r290509 built by root @ prigw01 on a i686 running Linux on 2010-11-30 22:12:05 UTC DAHDI: dahdi-linux-complete-2.4.0+2.4.0 LibPRI: libpri-1.4.11.5 Wanpipe: wanpipe-3.5.18 Kernel: Linux prigw01 2.6.32-24-generic #39-Ubuntu SMP Wed Jul 28 06:07:29 UTC 2010 i686 GNU/Linux The card does not have a hardware echo canceler. It should use MG2 as specified in DAHDI's system.conf: #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2010-12-08 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 #dchan=24 echocanceller=mg2,1-23 hardhdlc=24 And, from chan_dahdi.conf: ;Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 switchtype=national context=ldrouted group=1 echocancel=yes signalling=pri_net channel =1-23 Any thoughts, pointers, suggestions? The echo is horrible, please help me make it stop. :-) -- _ -- 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] Forking a call
Hi Mike, Fork will generated 2 CDRs, and will seperate CDRs But seems that there is a trouble in 1.6 (1.4 was working fine) For exemple : phone A (leg A) is called, I play some background sound and before putting in relation with phone B (leg B) I do a ForkCDR() in 1.4 billsec in the first CDRs whas the call time on leg A and in the second CDR the time of call on leg B in 1.6 billsec on the two CDRs is the call time on leg B in my case it cause some trouble since I cannot not charge the introduction message (background sound) Is it a known Bug ? or may be it's not a bug? regards Mickael 2010/9/23 Mike l...@net-wall.com Hi, Using 1.6.2.13. I'd like to know how I can force Asterisk to fork a call. To simplify things, Let's say I have an out context (for outbound calls) and an in (for inbound). If person A wants to call person B, and both are on my servers, I don`t want to send the call out. I want all this to happen internally on my server. The problem is if I use some condition to send calls in my out context back to my in context, some channel variables get mixed up, and (for example) when the calling part puts the called party on hold, the music on hold used is the called party's music. I am sure there are some less benign problems that could come with that. Is ForkCDR() what I am looking for? Any things I gotta watch out for when using it? I basically would like Asterisk to treat this call as two separate calls, as if one was completely outbound and the second an independant inbound call. Mike -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer45KSouth London
Wait, is 70k US for an experienced engineer supposed to be adequate? Thank you, not only that , but also note that it would be 70K at the US dollar exchange rate. However, because it is 45K Euros/Pounds earned and spent in UK, for all practical purposes it is just the same as if it was 45K US Dollars earned in the USA. On December 22, 2010 at 3:49 PM Watkins, Bradley bradley.watk...@compuware.com wrote: Wait, is 70k US for an experienced engineer supposed to be adequate? From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfDanny Nicholas Sent:Wednesday, December 22, 2010 2:27 PM To:'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London Wouldn't that be 70K USD? Or should we REALLY be worried about the British economy? From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfDon Kelly Sent:Wednesday, December 22, 2010 12:24 PM To:'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfC. Savinovich Sent:Wednesday, December 22, 2010 10:23 AM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net [mailto:ja...@langleyjames.co.uk] Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com--
Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London
By UK standards that's a pretty good salary. Bear in mind that there is no real 1:1 parity in IT salaries. In the US we earn significantly more for our IT efforts than in the UK. To give you an example, when I moved from London to New York I got a 4 fold pay rise in real terms for doing exactly the same job. I was on 28K GBP over there and got paid 120K US$ over here. On 12/22/2010 02:27 PM, Danny Nicholas wrote: Wouldn’t that be 70K USD? Or should we REALLY be worried about the British economy? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Don Kelly *Sent:* Wednesday, December 22, 2010 12:24 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It’s about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *C. Savinovich *Sent:* Wednesday, December 22, 2010 10:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: **Job Description: Asterisk MySQL Support Engineer** Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. *Required Skills Qualifications:* Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. *Kind Regards Jess* *08451249555* **Jess Hart*** *__* **Langley James IT Recruitment*** 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- /\/\ark Phillips -- _ -- 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] Vacancy - Asterisk MySQL Support Engineer45KSouth London
Lots of unemployed engineers in the US would be more than happy with 70K, or even less. A long period of high unemployment in the US, and world markets is something many have yet to come to understand. John Novack Watkins, Bradley wrote: Wait, is 70k US for an experienced engineer supposed to be adequate? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, December 22, 2010 2:27 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London Wouldn't that be 70K USD? Or should we REALLY be worried about the British economy? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Don Kelly *Sent:* Wednesday, December 22, 2010 12:24 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *C. Savinovich *Sent:* Wednesday, December 22, 2010 10:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: **Job Description: Asterisk MySQL Support Engineer** Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. *Required Skills Qualifications:* Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. *Kind Regards Jess* *08451249555* ** ** **Jess Hart*** *__* **Langley James IT Recruitment*** 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572
[asterisk-users] Maximum retries exceeded
have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 for seqno 1620 (Critical Response) Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging up call 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 no reply to our critical packet. -- Original -- From: asterisk-users-requestasterisk-users-requ...@lists.digium.com; Date: Thu, Dec 23, 2010 10:52 AM To: 229838677229838...@qq.com; Subject: confirm 7544ab1cb7b90cb7d6583f225fe16c45193d0c45 Mailing list subscription confirmation notice for mailing list asterisk-users We have received a request from 125.70.77.55 for subscription of your email address, 229838...@qq.com, to the asterisk-users@lists.digium.com mailing list. To confirm that you want to be added to this mailing list, simply reply to this message, keeping the Subject: header intact. Or visit this web page: http://lists.digium.com/mailman/confirm/asterisk-users/7544ab1cb7b90cb7d6583f225fe16c45193d0c45 Or include the following line -- and only the following line -- in a message to asterisk-users-requ...@lists.digium.com: confirm 7544ab1cb7b90cb7d6583f225fe16c45193d0c45 Note that simply sending a `reply' to this message should work from most mail readers, since that usually leaves the Subject: line in the right form (additional Re: text in the Subject: is okay). If you do not wish to be subscribed to this list, please simply disregard this message. If you think you are being maliciously subscribed to the list, or have any other questions, send them to asterisk-users-ow...@lists.digium.com.-- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
My h extension is in the same context as my Dial commands. Here is a cut
back version of the code.
The cause_code value is never stored in the mysql databae even when set in
the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = s,1,Macro(${ARG1})
exten = s,n,Set(CALLERID(name)=${siteDefaultCIDName})
exten = s,n,Set(CALLERID(number)=${siteDefaultCIDNumber})
exten = s,n,SipAddHeader(X-interNetGR-linetype:${gbl_ibclinetype})
exten = s,n,SipAddHeader(X-interNetGR-actlineid:${gbl_actlineid})
exten = s,n,Set(GROUP()=${siteGrpLineCount})
exten = s,n,Set(c_DialArg=${ARG2})
exten = s,n,Set(c_DialExten=${MACRO_EXTEN})
exten = s,n,GoSub(DoLineCountCheck,1)
exten = s,n,GotoIf($[${siteOverLineCount}=1]?OverLineCount,1)
exten = s,n,GosubIf($[${c_DialExten}=${siteDirSer}]?OverLineCount,1)
exten = s,n,GosubIf($[${c_DialExten}=411]?nofeature,1)
exten = s,n,GosubIf($[${siteUseE164}=1]?doDialE164,1:doDialStd,1)
exten = s,n,Goto(rc-${DIALSTATUS},1)
exten = s,n,Busy(60)
exten = s,n,Hangup()
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
exten = doDialStd,1,NoOp(Calling Using No E164)
exten =
doDialStd,n,Macro(OBD-CheckOutboundNumber,${c_DialArg}${c_DialExten})
exten =
doDialStd,n,Dial(${siteDefaultOutboundTrunk}/${c_DialArg}${c_DialExten},120,
ge${siteDialOptionsPublic})
exten = doDialStd,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = doDialStd,n,Return
exten = rc-ANSWER,1,NoOp(Do Return ANSWER)
exten = rc-ANSWER,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = rc-ANSWER,n,Hangup()
exten = rc-BUSY,1,NoOp(Do Return BUSY)
exten = rc-BUSY,n,Busy()
exten = rc-BUSY,n,Hangup()
exten = rc-NOANSWER,1,NoOp(Do Return NOANSWER)
exten = rc-NOANSWER,n,NoOp(Cause Code = ${HANGUPCAUSE})
exten = rc-NOANSWER,n,Hangup()
Any more feed back would be appercaited.
Bryant
From: Tilghman Lesher tilgh...@meg.abyt.es
Sent: Wednesday, December 22, 2010 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be ; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Sounds like your h extension is in the wrong context. Try including some
information about where you are putting the h extension and what
includes
you're doing.
--
Tilghman
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
No this is just a snip of the much larger code.
The h extension is runing but no values port dial function aer being written.
If I do a Set(CDR(field)=Value) before the dial
The value is stored. See my other response for a larger snip of code.
Bryant
From: Carlos Chavez cur...@telecomabmex.com
Sent: Wednesday, December 22, 2010 2:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some
values to the CDR then.
For the write of the CDR and stopping the billing seconds. Then
continue to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in
the h quickly store some values to the CDR then.For the write of the
CDR and stopping the billing seconds. Then continue to process some
cleanup funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Is the CDR line your only h line? I ask because if you only have one
priority for h then you MUST have:
exten = h,1,Set(CDR(cause_code)=${HANGUPCAUSE})
This is because the dialplan will not use n for the first priority and
thus will never run.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Simplifying dial-plan
On Wed, Dec 22, 2010 at 12:59 PM, Warren Selby wcse...@selbytech.com wrote:
On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese rsre...@gmail.com wrote:
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
Sorry, here's the rest.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
Why not make a Macro (or GoSub) to handle this block of code, and then your
outbound dial lines are just one line calling the Macro? Saves a lot of
repeating blocks of code. Something like this (not tested):
[macro-OutboundDial]
; ${ARG1} = CHANNEL
; ${ARG2} = EXTERNAL_CALLERID
exten = s,1,Set(Outgoing=${CUT(${ARG1},/,2)})
exten = s,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = s,n,GotoIf($[${Outgoing} = 201]?outbound2:outbound1)
exten = s,n(outbound1),Set(CALLERID(all)=${ARG2})
exten = s,n,Dial(SIP/${macro_ext...@vitel-outbound)
exten = s,n,Goto(h,1)
exten = s,n(outbound2),Set(CALLERID(all)=${ARG2})
exten = s,n,Dial(SIP/${macro_ext...@vitel-outbound2)
exten = s,n,Goto(h,1)
[outbound-context]
exten = _NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _NXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
Thanks Warren, that's what I'm looking to do.
First question is where did ${MACRO_EXTEN} come from, I assumed
${EXTEN} is a built in variable?
Secondly, where would the 1 and/or area-code need to be placed? Could
an additional argument be used to specify the prefix, i.e. a third
variable be specified in the outbond-context to implement the
OutboundDial macro, or is the MACRO_EXTEN suppose to be an
implementation of this?
exten = s,n,Dial(SIP/{$arg3}${macro_ext...@vitel-outbound2)
As Jeroen mentioned previously a goto may be used, would this help,
seems similar to what I am trying to accomplish.
exten = _NXXNXX,1,Goto(1${EXTEN},1)
exten = _NXX,1,Goto(1555${EXTEN},1)
Thanks,
Stephen
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Re: [asterisk-users] Simplifying dial-plan
On Wed, Dec 22, 2010 at 10:12 PM, Stephen Reese rsre...@gmail.com wrote:
Thanks Warren, that's what I'm looking to do.
First question is where did ${MACRO_EXTEN} come from, I assumed
${EXTEN} is a built in variable?
Secondly, where would the 1 and/or area-code need to be placed? Could
an additional argument be used to specify the prefix, i.e. a third
variable be specified in the outbond-context to implement the
OutboundDial macro, or is the MACRO_EXTEN suppose to be an
implementation of this?
exten = s,n,Dial(SIP/{$arg3}${macro_ext...@vitel-outbound2)
As Jeroen mentioned previously a goto may be used, would this help,
seems similar to what I am trying to accomplish.
exten = _NXXNXX,1,Goto(1${EXTEN},1)
exten = _NXX,1,Goto(1555${EXTEN},1)
To answer your first question - ${MACRO_EXTEN} is a macro-specific
variable. It's the ${EXTEN} that called the macro, since using ${EXTEN}
inside a Macro would just give you a value of s.
As for your second question, that's pretty easy to do. If every outbound
call needs to be formatted in the format 1NXXNXX, you would do this
(again, untested, but should be good along with the macro I gave you
earlier):
[globals]
DEFAULT_AREA_CODE=555 ; swap with your default area code
[outbound-context]
exten = _1NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _NXXNXX,1,Goto(outbound-context,1${EXTEN},1)
exten = _NXX,1,Goto(outbound-context,1${DEFAULT_AREA_CODE}${EXTEN},1)
exten = _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = 911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same context as my Dial commands. Here is a
cut back version of the code.
The cause_code value is never stored in the mysql databae even when set
in the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
There's the problem. The h extension should be in whatever context is
calling the Macro, not in the Macro context itself.
--
Tilghman
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[asterisk-users] Asterisk handling multiple simultaneous calls for IVR
Hi I am new to asterisk. I want to build an IVR system so that approximately 10-15 users can call simultaneously and use the same dialplan.. We have PRI lines and are thinking of buying Digium TE!21 card for my software.Would it serve my needs? Please let me know how to configure dialpan and other configuration plans for my problem. And can I simulate it without using a hartware before buying it? Bhavesh -- _ -- 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] Maximum E1 Ports on Asterisk ?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Wednesday, December 22, 2010 7:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, December 22, 2010 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ? On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo zoelha...@yahoo.co.id wrote: Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH Zoel It is possible to do what you are asking. In general the issue is raised about having all your eggs in one basket where one server or hardware failure can drop all of your lines for a period of time. External solutions like Xorcom and Redfone are great ways of abstraction. The concurrent call load on a server relies on the work to be done on each call. If you are using multiple codecs and recording the calls in another file format with other complex dialplan or AGI scripts then one server may not handle the calls well. If everything is ALAW and just dialing though then this would not be a problem for one server. If you search the list for sizing concurrent and load you will find more information. One very nice thing is that testing is very easy with or without the E1 hardware, try running the TDMoE channels between two servers and run a SIPp or other test to see the issues in a lab. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ In a previous post I also mentioned Xorcom. They do have a unique fail over ability with their Astribank systems. With dual servers, separate chassis and power supplies for the 4 port T1/E1 cards, USB interconnections, and redundant power supplies for the Astribanks, system downtime can be minimized, and if there is a failure, repair would be at worst, no screwdriver needed. If system failure would be idling 200 - 400 people, avoiding system down time would be a major objective. Cary Fitch Thanks Cary and Andrew, This is a great suggestion and alternative for me. I will take a look at Xorcom and the AstriBank. Once again, Thanks guys. ZH -- _ -- 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] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
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Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can
see, for the other types of reasons, the DIALSTATUS got a value (and we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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[asterisk-users] Forward voicemail to group of people
Aloha, Is there a way to forward a message to multiple people from within the telephone user interface? Now there is only the ability to forward to an individual. I see there is a way to leave a message for multiple people using the dial plan but that is not available when you are listening to voicemail. Thanks! Matt -- _ -- 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
