Re: [asterisk-users] Asterisk Error
This means that no ethernet interface is found for seeding the global EID. So you will have to set it manually. :) Pretty clear. On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do Load-Balancing for Asterisk with OpenSIPS
Great Job Bogdan On Tue, Mar 10, 2009 at 12:52 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi, When trying to cluster Asterisk boxes to gain scalability and more performance, there is now a new simple and efficient solution for doing it. OpenSIPS/OpenSER 1.5 can now implement traffic routing based on load. Shortly, when OpenSIPS routes calls to a set of destinations, it is able to keep the load status (as number of ongoing calls) of each destination and to choose to route to the less loaded destination (at that moment). OpenSIPS is aware of the capacity of each destination - it is preconfigured with the maximum load accepted by the destinations. This is an idea Load-Balancer to front your Asterisk cluster. A nice tutorial about how to do LB for your Asterisk is available: http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Regards, Bogdan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk On Solaris Real Time
Hi All I got Asterisk to run on Solaris however I do need it to run in realtime mode I.e. with the res_mysql file. Did anyone succeed in this ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk On Solaris
Hi All I got Asterisk to run on Solaris however I do need it to run in realtime mode I.e. with the res_mysql file. Did anyone succeed in this ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail in Real Time
Hi I do have asterisk running in real time I do want to add voicemail to real time. I did follow : http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail However when I do try to make a voicemail I do get : [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! -- Executing [EMAIL PROTECTED]:1] VoiceMail(SIP/alijawad-08aaf0f0, [EMAIL PROTECTED]|u) in new stack [Nov 20 12:17:08] WARNING[22277]: app_voicemail.c:2862 leave_voicemail: No entry in voicemail config file for '999alijawad' -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/alijawad-08aaf0f0, ) in new stack == Spawn extension (a2billing, 999alijawad, 2) exited non-zero on 'SIP/alijawad-08aaf0f0' Even though I do have 999alijawad as a mailbox in voicemail_users If I do setup the mailbox in voicemail.conf it works fine. [a2billing] 999alijawad = 123456, alijawad, [EMAIL PROTECTED] I did setup extconfig.conf as it should be: voicemail = mysql,mya2billing,voicemail_users Please advice ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ERROR:Failed to create H323 listener
Hi I am trying to get H323 to run on Asterisk, basically I had Asterisk running so I followed this tutorial http://astrecipes.net/index.php?n=286 and got h323 to run on my first server on the second server it is just throwing the error: ERROR:Failed to create H323 listener The whole error is : ERROR: Could not open H.323 listener port on 1720 [Oct 13 09:20:48] ERROR[3608]: chan_h323.c:3166 load_module: Unable to create H323 listener. From /var/log/asteris/h323 09:14:43:478 Error:Bind failed 09:14:43:478 ERROR:Failed to create H323 listener 09:14:43:478 Destroying H323 Endpoint I have checked and nothing is running on 1720, I even tried other ports Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get Call Length of Calls
Hi I am using show cannels verbose to get info about my current sip calls. However, the time displayed is always zero. Any hints ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get Call Length of Calls
Hi Thanks for the hint, however I do already have a cdr tool for finished calls. core show channels verbose does show the duration of calls in real time. However, it does not work all the time, I.e. at times it works great other times it just displays 0 for the call duration, although the call is up and running. On Fri, Sep 26, 2008 at 4:37 PM, Julien Claassen [EMAIL PROTECTED] wrote: i! Not about this directly, but an alternative. If you need the length of finished calls, work with the system. Use a specific call to the date command, so it's easy to evaluate the time info or some other tool to give you an absolute of time. Then at the end of the call use another system call to that program and subtract the two values. You could use bc for this. It even works with decimal numbers. One or two shell-scripts will do this trick. HTH. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- With Regards Ali Jawad System Administrator http://www.alijawad.org Phone : +961-01-559031 Mobile : +961-03-041705 Confidentiality Notice: The contents of this E-mail are intended for the named recipient only. It may contain confidential and privileged information. If you received it in error, please notify us immediately and then destroy it. Internet communications are not secure and therefore I do we do not accept legal responsibility for the contents of this message. Also, and though we provide every effort to keep our network free from viruses, you would need to check this E-mail and any attachments for viruses as we can take no responsibility for any computer virus which might be transferred by way of this E-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Found unknown media description format
Hi One of my softphones is supposed to support g711 , however I am getting these errors and a 404 not found when I try to make a call from it. However on xlite it works fine using g711. Below is the log of the phone that is not working. Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0 a=ice-pwd:00gnam9Pd+SG6KzQNLf1fS a=ice-ufrag:Xng7 m=audio 19504 RTP/AVP 103 18 102 0 8 97 119 117 100 101 13 105 106 a=rtcp:19505 a=candidate:4 1 UDP 2122300927 192.168.0.176 19504 a=candidate:1 1 UDP 2122285311 169.254.2.2 19504 a=candidate:2 1 UDP 2122285055 192.168.238.1 19504 a=candidate:3 1 UDP 2122284799 192.168.111.1 19504 a=candidate:5 1 UDP 1694482431 193.227.186.146 19504 a=candidate:6 1 UDP 1677 87.236.144.70 41343 a=candidate:4 2 UDP 2122300926 192.168.0.176 19505 a=candidate:1 2 UDP 2122285310 169.254.2.2 19505 a=candidate:2 2 UDP 2122285054 192.168.238.1 19505 a=candidate:3 2 UDP 2122284798 192.168.111.1 19505 a=candidate:5 2 UDP 1694482430 193.227.186.146 19505 a=candidate:6 2 UDP 1676 87.236.144.70 41344 a=rtpmap:103 ISAC/16000 a=fmtp:18 annexb=no a=rtpmap:102 iLBC/8000 a=rtpmap:97 IPCMWB/16000 a=rtpmap:119 ISACLC/16000 a=rtpmap:117 red/8000 a=rtpmap:100 EG711U/8000 a=rtpmap:101 EG711A/8000 a=rtpmap:105 CN/16000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=sendrecv - --- (16 headers 33 lines) --- Sending to 87.236.144.9 : 5060 (no NAT) Using INVITE request as basis request - 0c49de60-1f17-4de8-aa0b-ae3f7b7527b9 Found no matching peer or user for 'xx.xx.xx.xx:5060' Found RTP audio format 103 Found RTP audio format 18 Found RTP audio format 102 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 119 Found RTP audio format 117 Found RTP audio format 100 Found RTP audio format 101 Found RTP audio format 13 Found RTP audio format 105 Found RTP audio format 106 Peer audio RTP is at port 192.168.0.176:19504 Found unknown media description format ISAC for ID 103 Found audio description format iLBC for ID 102 Found unknown media description format IPCMWB for ID 97 Found unknown media description format ISACLC for ID 119 Found unknown media description format red for ID 117 Found unknown media description format EG711U for ID 100 Found unknown media description format EG711A for ID 101 Found audio description format CN for ID 105 Found audio description format telephone-event for ID 106 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.176:19504 Looking for 5678 in default (domain ser..net) Any ideas ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk With Web meetme
Hi Dan I got localhost*CLI cb mysql status No such command 'cb mysql' (type 'help' for help) Asterisk 1.4 and Meetme is the latest version 3.0, ztdummy is working fine. Thanks On Thu, Jun 26, 2008 at 6:48 PM, Dan Austin [EMAIL PROTECTED] wrote: Ali wrote: I followed this howto http://www.voip-info.org/wiki/view/MeetMe-Web-Control and http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html to install web meetme with asterisk, I know the meetme module is included however I need to be able to ban and mute users as well. All of the installation went fine however when I do call a conference number I create using the interface all I get is service unavailable, I did run asterisk in verbose mode that did not make me any smarter. I added to extensions.conf the following [confserv] ;Make sure you change 1199 to your conference bridge extension(s) ;more information on this can be found at the asterisk web site. exten = 121212,1,Answer exten = 121212,n,Wait(3) exten = 121212,n,CBMysql() exten = 121212,n,Hangup Where 121212 is an existing extension, I really dont get it this all of the documentation available but I surely missed something here..any hints please ? Let's start with the easy stuff, if confserv included in the context that the phone has access to? What is the output of the command CLIcb mysql status? What version of Asterisk and Web-MeetMe are you using? Do you have a timing source (ztdummy or PSTN interface card)? Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- With Regards Ali Jawad System Administrator http://www.alijawad.org Phone : +961-01-559031 Mobile : +961-03-041705 Confidentiality Notice: The contents of this E-mail are intended for the named recipient only. It may contain confidential and privileged information. If you received it in error, please notify us immediately and then destroy it. Internet communications are not secure and therefore I do we do not accept legal responsibility for the contents of this message. Also, and though we provide every effort to keep our network free from viruses, you would need to check this E-mail and any attachments for viruses as we can take no responsibility for any computer virus which might be transferred by way of this E-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk With Web meetme
Hi I followed this howto http://www.voip-info.org/wiki/view/MeetMe-Web-Control and http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html to install web meetme with asterisk, I know the meetme module is included however I need to be able to ban and mute users as well. All of the installation went fine however when I do call a conference number I create using the interface all I get is service unavailable, I did run asterisk in verbose mode that did not make me any smarter. I added to extensions.conf the following [confserv] ;Make sure you change 1199 to your conference bridge extension(s) ;more information on this can be found at the asterisk web site. exten = 121212,1,Answer exten = 121212,n,Wait(3) exten = 121212,n,CBMysql() exten = 121212,n,Hangup Where 121212 is an existing extension, I really dont get it this all of the documentation available but I surely missed something here..any hints please ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parsing incoming extension till first @
Thanks for the hint Patrick I appreciate it.
On Tue, Apr 22, 2008 at 3:02 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Using _. is going to result in warnings. A much better practice is to
use _X.
Ali Jawad wrote:
Thx again patrick it worked, I used
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,1,Dial(SIP/[EMAIL PROTECTED])
while it should have been
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,2,Dial(SIP/[EMAIL PROTECTED])
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!DSPAM:480e5326213018190740810!
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Ali Jawad System Administrator
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Phone : +961-01-559031
Mobile : +961-03-041705
Confidentiality Notice: The contents of this E-mail are intended for the
named recipient only. It may contain confidential and privileged
information. If you received it in error, please notify us immediately and
then destroy it. Internet communications are not secure and therefore
I do we do not accept legal responsibility
for the contents of this message. Also, and though we provide every effort
to keep our network free from viruses, you would need to check this E-mail
and any attachments for viruses as we can take no responsibility for any
computer virus which might be transferred by way of this E-mail.
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[asterisk-users] Parsing incoming extension till first @
Hi All
When I dial a number it reaches the asterisk switch as [EMAIL PROTECTED]@123.com
what I need to do is to parse the abc and send it to my pstn gateway
as in
exten = _.,2,Dial(SIP/[EMAIL PROTECTED])
this does work but I do have a varying number of numbers before the @
exten = _.,1,Dial(SIP/${EXTEN:0:[EMAIL PROTECTED])
Well can I use some kind of regular expression to take all numbers
before the first @ and send them to the pstn
something like
exten = _.,1,Dial(SIP/${regexp(condition,Exten)[EMAIL PROTECTED])
thx
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Re: [asterisk-users] Parsing incoming extension till first @
Thanks Patrick this resulted in
-- Executing [EMAIL PROTECTED]@google-in:1]
Set(Gtalk/jabber1-c06e, dst=009613041705) in new stack
-- Auto fallthrough, channel 'Gtalk/jabber1-c06e' status is 'UNKNOWN'
It seems to have cut the correct part but I am not sure about the rest
of it, it is causing auto fallthrough with status UNKNOWN
On Tue, Apr 22, 2008 at 1:33 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Ali Jawad schrieb:
When I dial a number it reaches the asterisk switch as [EMAIL
PROTECTED]@123.com
what I need to do is to parse the abc and send it to my pstn gateway
as in
exten = _.,2,Dial(SIP/[EMAIL PROTECTED])
this does work but I do have a varying number of numbers before the @
exten = _.,1,Dial(SIP/${EXTEN:0:[EMAIL PROTECTED])
Well can I use some kind of regular expression to take all numbers
before the first @ and send them to the pstn
something like
exten = _.,1,Dial(SIP/${regexp(condition,Exten)[EMAIL PROTECTED])
core show function CUT
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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With Regards
Ali Jawad System Administrator
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Phone : +961-01-559031
Mobile : +961-03-041705
Confidentiality Notice: The contents of this E-mail are intended for the
named recipient only. It may contain confidential and privileged
information. If you received it in error, please notify us immediately and
then destroy it. Internet communications are not secure and therefore
I do we do not accept legal responsibility
for the contents of this message. Also, and though we provide every effort
to keep our network free from viruses, you would need to check this E-mail
and any attachments for viruses as we can take no responsibility for any
computer virus which might be transferred by way of this E-mail.
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Re: [asterisk-users] Parsing incoming extension till first @
Thx again patrick it worked, I used
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,1,Dial(SIP/[EMAIL PROTECTED])
while it should have been
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,2,Dial(SIP/[EMAIL PROTECTED])
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[asterisk-users] Asterisk Jingle-SIP GW Question
Dear All I am using gtalk features with my own XMPP server OpenFire I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls from clients registered on my XMPP server to SIP devices by calling the xmpp accounts registered as clients on asterisk. So far so good. So if I want to call sip:1000 I call the xmpp account that is bound to that account in extensions.conf. However what do I have to do to make this work with PSTN numbers. I can just setup an entry + extensions for each pstn number I want to call. I know that I can parse the incoming number and send it to the PSTN with sip, however with jingle the number must be online already since jingle is presence based. So I must have a registered client for each number I want to call in the following format XMPP --- SIP 1000 To Call 1000 //sip extension 1001 To Call 15461315461 //pstn num 1002 To Call 46456543213 //cell phone num So in essence I need to have one entry in jabber.conf per number, is there something dynamic that can be done ? Thanks asterisk-users@lists.digium.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UPDATED Asterisk Jingle Extensions.conf
Dear All I am using gtalk features with my own XMPP server OpenFire I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls from clients registered on my XMPP server to SIP devices by calling the xmpp accounts registered as clients on asterisk. I have sent a previous email with a problem that I solved by using component mode. In this mode the asterisk server acts as a subdomain. So I can call [EMAIL PROTECTED], [EMAIL PROTECTED] My current extension file looks as follows: [google-in] exten = s,1,NoOp( Call from XMPP) exten = s,n,Set(CALLERID(name)=From XMPP Server) exten = s,n,Dial(SIP/1234) However I want it to call the number in dialed initially I.e 1000 or 1001 etc etc etc. Any way to do this parsing using Asterisk ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jingle with Asterisk + PSTN
So should I register directly on the asterisk server or should I send the voice calls through ejabberd to asterisk ? On Mon, Mar 31, 2008 at 4:55 PM, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Ali, On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote: Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Asterisk speaks Jingle indeed. If you're using a libjingle based client, you'll have to set up a GoogleTalk connection to Asterisk through the chan_gtalk channel driver. Those good pointers will help : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk http://taug.ca/node/43 Note : the chan_jingle channel driver implements the Jingle (not GoogleTalk related), so it won't work with libjingle even though the names sound close. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN ? PSTN, SIP, H323, SCCP, ... or any protocol implemented in Asterisk! Philippe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jingle with Asterisk + PSTN
Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN ? Thx Any feedback is appreciated. Note: I do not intend to implement SIP in my client ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
