Re: [asterisk-users] CDR with unix time.
Just use uniqueid, which is exactly what you want. No modification is necessary. Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook http://www.facebook.com/pages/Vilnius-Lithuania/Kolmisoft/106746839379147 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang Sent: Thursday, February 10, 2011 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDR with unix time. Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site http://openingyourmind.wordpress.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] usage of account code in CDR
We use it to determine who is the caller. Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Wednesday, November 24, 2010 6:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] usage of account code in CDR please reply on this if u know On 11/18/2010 09:24 AM, Nikhil wrote: Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil -- _ -- 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] Is existing CDR in Asterisk is enough for complete billing
From our experience it is not enough. We had to rewrite CDR generation to suite our billing needs. That was on 1.4.xx, we are not using 1.6+ Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Monday, November 22, 2010 7:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil -- _ -- 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] peer IP address in CDR
For codecs use CHANNEL function, but you will only get CallLegA codecs.
Without hacking Asterisk, you will not be able to get CallLegB codecs.
Patch for Asterisk 1.4.33.1 attached to get such info.
Retrieve such info with variables:
RTPAUDIOQOS
BRTPAUDIOQOS
And even more:
LEG1DATA
LEG2DATA
In format:
uniqueid|accountcode|chan_type|audionativeformat|audioreadformat|audiowritef
ormat|language|hangupcause|peerip|recvip|from|uri|useragent|
example:
LEG2DATA:
1277817284.0|7|SIP|alaw|alaw|alaw|en|16|192.168.0.148|192.168.0.148|sip:1003
@173test|sip:1...@192.168.0.148:5061|X-PRO build 1082
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Tuesday, June 29, 2010 6:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] peer IP address in CDR
Hi!
Do you already have script to capture user's IP address? If not, check
it here how I am capturing it:
http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-
within-the-dialplan
Or simply use one fo these:
${SIPCHANINFO(peerip)}
${SIPCHANINFO(recvip)}
${SIPCHANINFO(uri)}
More details with show function SIPCHANINFO on the CLI.
But: Anyone has an idea how to store the codec(s) that were employed for
the call in the CDR (or access it during hangup in the dialplan)?
The Wiki has a suggested patch to enhance SIPCHANINFO, but I wonder if
there is a cleaner and built-in way to do it:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
Philipp
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chan_sip.c.patch
Description: Binary data
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Re: [asterisk-users] Codec negotiation
Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: Tuesday, June 29, 2010 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Codec negotiation On 26 June 2010 22:08, Ryan Wagoner rswago...@gmail.com wrote: I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles. If Asterisk knows the trunk only supports ulaw why would it offer g722 to the phone. Ryan Because the codec is already chosen before the call is made, and you told it that g722 is permitted. There are all sorts of discussions in play about codec negotiation, but at this point in time, if you want different behaviour you'll need to go and code it yourself, and cross-channeltype this is not going to be trivial :) 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 -- _ -- 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] bug in asterisk
Execute such commands with cronjob every night: /etc/init.d/asterisk stop sleep 3 killall -9 safe_asterisk killall -9 asterisk /etc/init.d/asterisk start Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of benoit bernard Sent: Tuesday, May 11, 2010 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] bug in asterisk Hello all, i have asterisk installed in our call centre and we work 24h in day with this server ,the problem is each day in the night the server hangs and the calls stopped And i must to restart asterisk with this command “service asterisk restart” When i make service asterisk start i got the message failed that is mean that the service is already ON Any help please -- _ -- 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] bug in asterisk
Check Asterisk changelog (http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.31) with ctrl+f deadlock. Guess how many deadlock related bugs wonderful Digium programmers will solve in future releases? My proposition is not solution to the problem, its the survival guide in Asterisk world. We have 470 servers deployed around the world with Asterisk and this piece of code extended my and mine coworkers lifes by many years. If you really want to solve your problem - start here: http://www.voip-info.org/wiki/view/Asterisk+debugging Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vardan Sent: Tuesday, May 11, 2010 2:53 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] bug in asterisk Hello I think this is not right way :) Look the log's files, find the problem and resolv The cronjob is the way to stay fat always online, until you find the problem :) Vardan Mindaugas Kezys wrote: Execute such commands with cronjob every night: /etc/init.d/asterisk stop sleep 3 killall -9 safe_asterisk killall -9 asterisk /etc/init.d/asterisk start Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of benoit bernard Sent: Tuesday, May 11, 2010 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] bug in asterisk Hello all, i have asterisk installed in our call centre and we work 24h in day with this server ,the problem is each day in the night the server hangs and the calls stopped And i must to restart asterisk with this command service asterisk restart When i make service asterisk start i got the message failed that is mean that the service is already ON Any help please -- _ -- 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] Calculating R Factor and MOS metrics for VoIP
There is no problem to get necessary data for MOS calculation for Call Leg
A using:
${CHANNEL(rtpqos,audio,all)}
How to get similar data for Call Leg B?
It would be very nice to have such info even if it will not lead to correct
MOS calculation.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Monday, March 08, 2010 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calculating R Factor and MOS metrics for VoIP
Steve Underwood wrote:
MOS and R factor are the two QoS parameters used to estimate VoIP call
quality.
You can't calculate MOS. Its an assessment based on a lot of human
hearing. Nonetheless, there is a profitable industry in pretending to
calculate MOS.
Right, computers don't have 'opinions' (the 'O' in 'MOS'). However, it
seems that many people use PESQ scores as a MOS-equivalent for test and
planning purposes now. However, that requires running predefined samples
through the system under test, not just calculations based on network
effects of real calls.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] rtcachefriends qualify sip reload
From my experience prune does not take effect without reload. And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, March 02, 2010 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends qualify sip reload On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote: On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: If you are changing RealTime config in your DB you need to do a sip prune realtime either directly from asterisk cli or using AMI. You really do not need to do a SIP reload when changing the config of one sip extension. I notice that after a sip prune realtime all I also loose all of my realtime sip peers. Same result actually as with sip reload. I close the softphone of gerrie2 (becomes unspecified) asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie005/gerrie005192.168.1.106D N 5060 OK (4 ms) Cached RT gerrie002/gerrie002(Unspecified)D N 0 UNKNOWNCached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT I prune the realtime peers to no longer have gerrie002 in cache : asterisk*CLI sip prune realtime all 3 peers pruned. 2 users pruned. [Mar 2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 91 The realtime peers are all gone : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime Internal call fails : [Mar 2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Mar 2 15:46:38] == Everyone is busy/congested at this time (1:0/0/1) [Mar 2 15:46:38] == Auto fallthrough, channel 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL' I re-register 2 softphones (gerrie001 gerrie005) : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002(Unspecified)D N 0 UNREACHABLE Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT gerrie005/gerrie005192.168.1.106D N 5060 OK (7 ms) Cached RT The SIP-peer 'gerrie002' is still in the cache ! Don't know where this is coming from ?? I prune again : asterisk*CLI sip prune realtime all 3 peers pruned. 1 users pruned. [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 And again no more peers until I re-register : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime This realtime thing isn't really working out here... What exactly do I need to do to clear the cache and thus the old SIP-peers so they can no longer be used ?? Do not prune all peers, only the peer you wish to reload or eliminate! Do sip prune realtime peer peername. That way you do not lose all the other registrations. I really do not see this as a problem as the phones will usually re register quickly or if the user dials any number. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] rtcachefriends qualify
The problems we have with Asterisk Realtime: 1. After reload all registrations are void. 2. Without reload prune does not take effect. Test it in your scenario also. Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, March 02, 2010 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends qualify Thank you for your answer, Nic. It seems that by putting rtcachefriend=yes, the qualify works as expected and even changes made to my realtime MySQL-DB take affect immediately without the need of a reload (I changed the username and name). However the old username and name are still valuable and using this old SIP-user, one can still make outgoing calls. Receiving calls is no longer possible : WARNING[32439]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Adding 'rtautoclear=yes' to sip.conf makes no difference. Changes to SIP-account are taken immediately, but the old SIP-credentials are still valid. (even after an unregister and re-register) Only after a sip reload I get the notice : [Mar 2 10:41:03] NOTICE[32498]: chan_sip.c:15889 handle_request_register: Registration from 'Gerriesip:gerrie0...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found So a sip reload is always necessary to clear the cache ?? Jonas. On Mon, 2010-03-01 at 14:31 +, Nic Colledge wrote: Hi, I think so, maybe someone can help clarify this for me also. I have: rtcachefriends=yes rtautoclear=yes in sip.conf and was under the impression that this caches the settings from the database until a user unregisters. When they unregister the data is removed from the cache (rtautoclear). For me this was a nice compromise. This is from memory but I’m pretty sure I got this from the documentation online, if someone can confirm what I’m saying that would be sweet. Thanks. Nic. -- _ -- 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] rtcachefriends qualify sip reload
Sip reload Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: mailto:i...@kolmisoft.com i...@kolmisoft.com URL: http://www.kolmisoft.com http://www.kolmisoft.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, March 02, 2010 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends qualify sip reload On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: If you are changing RealTime config in your DB you need to do a sip prune realtime either directly from asterisk cli or using AMI. You really do not need to do a SIP reload when changing the config of one sip extension. I notice that after a sip prune realtime all I also loose all of my realtime sip peers. Same result actually as with sip reload. I close the softphone of gerrie2 (becomes unspecified) asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie005/gerrie005192.168.1.106D N 5060 OK (4 ms) Cached RT gerrie002/gerrie002(Unspecified)D N 0UNKNOWN Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT I prune the realtime peers to no longer have gerrie002 in cache : asterisk*CLI sip prune realtime all 3 peers pruned. 2 users pruned. [Mar 2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 91 The realtime peers are all gone : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime Internal call fails : [Mar 2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Mar 2 15:46:38] == Everyone is busy/congested at this time (1:0/0/1) [Mar 2 15:46:38] == Auto fallthrough, channel 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL' I re-register 2 softphones (gerrie001 gerrie005) : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002(Unspecified)D N 0UNREACHABLE Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT gerrie005/gerrie005192.168.1.106D N 5060 OK (7 ms) Cached RT The SIP-peer 'gerrie002' is still in the cache ! Don't know where this is coming from ?? I prune again : asterisk*CLI sip prune realtime all 3 peers pruned. 1 users pruned. [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 And again no more peers until I re-register : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime This realtime thing isn't really working out here... What exactly do I need to do to clear the cache and thus the old SIP-peers so they can no longer be used ?? Jonas. -- _ -- 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] A2Billing and other prepaid Billing like ASTCC, who is better?
Please try our billing which has easier managing interface and works ok with H323: http://www.voip-info.org/wiki/view/MOR FREE version is available over this link: http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/ Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: 2010 m. vasario 7 d. 01:20 To: asterisk-users@lists.digium.com Subject: [asterisk-users] A2Billing and other prepaid Billing like ASTCC, who is better? Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with Asterisk and Gnugk at the same time (instead of using one billing for asterisk and one billing for gnugk, specially that gnugk is good for h323 functionalities that are missing in asterisk). Appreciate any help and advise in that direction. Regards Bilal -- _ -- 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] sip realtime md5secret
Just remember, that after reload you will lose all registrations. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: 2010 m. vasario 2 d. 22:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip realtime md5secret On Tue, 2010-02-02 at 21:20 +0200, Emre Kurnaz wrote: Hi all, Does asterisk cache realtime sip md5secret values? I create a user over a web site and set a password as asd and I can login with that password. After a while I change my password and set it as 123. Although the password is set as 123 in the mysql database (I double checked), i can not login using the password 123, but with asd. So, am i missing a point? or is this how asterisk works? and Should I reload asterisk after adding a peer in the database? Any help would be appreciated. If you have rtcachefriends=yes set in your sip.conf file then you either have to wait until the peer expires or you have to reload sip so the peer is re read from the database. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] How to get LEG B channel info?
Hello,
How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends?
I can use Dial G option to go to Leb B channel when call is answered, but
how to go here when call ends?
Is here any option/function in Dial Plan?
Or should I use ast_bridged_channel(chan) to get bridged channel and try to
retrieve data I need from internal structures using custom c module and
Asterisk API?
I'm trying to retrieve ${CHANNEL(rtpqos,audio,all)} for Leg B.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
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Re: [asterisk-users] Realtime SIP Register
Use #exec directive to execute external script which retrieves registration data from DB, and outputs correct registration string as text. Do not forget to enable #exec in asterisk.conf You will need to do sip reload each time you change registration settings. With reload you will lose all existing registrations and all previously registered devices will be unreachable till they register again. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Roos [Inlogia GmbH] Sent: 2009 m. lapkričio 27 d. 14:11 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Realtime SIP Register Hi, I would like to have my register directives from sip.conf in my mysql database: register = user[:secret[:authuse...@host[:port][/extension] I already have the sip users and the other config in the DB but how to get the register in there, too? In an old mail (Mon Oct 3 00:49:15 MST 2005) Olle E. Johansson said the [general] section can only be static. Has there anything changed in the last 4 years? Thanks! Philipp ___ -- 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] Scheduling destruction of SIP dialog
Hello, I got situation which is unclear for me, hope somebody could explain this. A calls to B INVITE sent from A to B B responds with 100 Trying B responds with 183 Progress After 10 seconds: Asterisk CLI: Scheduling destruction of SIP dialog '..' in 32000 ms (Method: INVITE) Asterisk sends CANCEL _instantly_ B responds with 200 OK and 487 Request Terminated Asterisk confirms 102 ACK CLI: Really destroying SIP dialog '..' Method: INVITE Call terminates Asterisk version 1.4.18.1 Total call duration: 11s Timeout on call to B is set to 60 seconds: 'SIP/0277027277...@prov7|60|S(7197)' Call log is here: http://pastebin.ca/1667975 Why Asterisk decided to terminate the call? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Billing applications
You can try free version of MOR Softswitch with billing and routing: http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/ We rewrote Asterisk CDR completely and yes, it supports transfers. More info about MOR: http://www.voip-info.org/wiki/view/MOR Free version supports up to 10 simultaneous calls which is enough for majority of startups. You can check our manual to see what functionality is supported: http://wiki.kolmisoft.com/index.php/MOR_Manual Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy Sent: 2009 m. spalio 9 d. 18:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Billing applications Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MPG123 Dying
Try: killall -9 safe_asterisk killall -9 asterisk /etc/init.d/asterisk start Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: 2009 m. spalio 6 d. 23:02 To: asterisk-users@lists.digium.com Subject: [asterisk-users] MPG123 Dying Please how do I stop the following ??? Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. mpg123: no process killed Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free version of softswitch with billing and routing released
Hello, We are happy to announce that FREE version of MOR 8 - our advanced Softswitch with billing and Routing is released. It comes as ISO image which installs everything from scratch. FREE edition has all functionality just limited to 10 simultaneous calls. We hope it will be useful for starters and makes life easier for many people. Link to get it: http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/ More info about software: http://www.voip-info.org/wiki/view/MOR Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choose IAX or SIP trunking?
We had many problems with IAX2, changing to SIP solved them all. Let me paste link to wise-words which clearly illustrates our experience: http://wiki.kolmisoft.com/index.php/Why_we_do_not_suggest_to_use_IAX2 Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kirill 'Big K' Katsnelson Sent: 2009 m. spalio 1 d. 02:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Choose IAX or SIP trunking? Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Thank you for answer. It was very informative, I put it in our wiki if you don't mind. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 2009 m. rugsėjo 22 d. 20:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTPAUDIOQOS On Tuesday 22 September 2009 10:42:44 Johann Steinwendtner wrote: Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txji tter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? It's a ratio of out-of-order (jittered) to in-order packets, calculated progressively. Due to the progressive calculation, it's not exactly 3/147, in this case, but it's close enough to know that 3 packets were received out-of-order. The closer the value is to 0, the better. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitt er=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
Asterisk sometimes goes to sleep. (And never wakes-up). Restart it and all will be fine again. We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does not respond – restarts it. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader Sent: 2009 m. rugsėjo 8 d. 10:40 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk CLI commands not running ! Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CHANNEL(rtpqos,audio,all)}
From my research:
Our Receiver
ssrc our ssrc
rxcount no. received packets/Received packets
lplost packets/Lost packets
rxjitter our calculated jitter(rx)/Jitter
Our Sender
themssrc their ssrc
txcount transmitted packets/Sent packet
rlp remote lost packets/Lost packets
txjitter reported jitter of the other end/Jitter
rtt round trip time/RTT
Synchronization source (SSRC): The source of a stream of RTP
packets, identified by a 32-bit numeric SSRC identifier carried in
the RTP header so as not to be dependent upon the network address.
All packets from a synchronization source form part of the same
timing and sequence number space, so a receiver groups packets by
synchronization source for playback. Examples of synchronization
sources include the sender of a stream of packets derived from a
signal source such as a microphone or a camera, or an RTP mixer
(see below). A synchronization source may change its data format,
e.g., audio encoding, over time. The SSRC identifier is a
randomly chosen value meant to be globally unique within a
particular RTP session (see Section 8). A participant need not
use the same SSRC identifier for all the RTP sessions in a
multimedia session; the binding of the SSRC identifiers is
provided through RTCP (see Section 6.5.1). If a participant
generates multiple streams in one RTP session, for example from
separate video cameras, each MUST be identified as a different
SSRC.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hh174
Sent: 2009 m. rugsėjo 5 d. 17:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ${CHANNEL(rtpqos,audio,all)}
Hi all,
With Asterisk 1.6.1.6
Trying to have statistic concerning Rtp audio quality, I use
${CHANNEL(rtpqos,audio,all)}
having also tried AUDIORTPQOS and ${CHANNEL(rtpqos,audio,...)}
Sometimes, it works and I have results.
Most of the time I get strange or no results even when the call was
succesfull.
rtpdest set at 0.0.0.0:0, no Joitter information, no packetlosts,...
It seems that when the channel is hungup, some informations are lost
(often the cas with rtpdest) depending on the party hanging-up.
Also, some info are not clear for me, like what are the meaning of
-rtt? (Delay?)
-ssrc=1271016709 (what is the meaning of this number?
-themssrc
Any clue, docs, informations to make the rtp statistics working?
What do I wrong?
Olivier
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Re: [asterisk-users] mysql sip realtime
If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime peername from the CLI Without reload prune does not take effect in 1.4.x And after reload all registrations are lost. So basically Asterisk Realtime is big mess from our experience and is totally unusable. We ended making #exec based script which takes data from DB and forms static configuration on each reload. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpjūčio 20 d. 15:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mysql sip realtime Hi The column order in your mysql sip table is irrelevant (Example sip table here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) All generic parameters are still taken from sip.conf and you must set rtcachefriends=yes If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime peername from the CLI As stated previously, you should never have to reload the sip module once realtime is working properly Hope this all helps Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mysql sip realtime
We use 1.4.18.1 and 1.4.26.1 and it does not work - settings are not changed after prune, asterisk must be reloaded, sip reload or iax2 reload makes changes. But after that all devices loose registration. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpjūčio 21 d. 12:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mysql sip realtime I have to disagree with you there, we use 1.4.17 and sip prune realtime works fine Mindaugas Kezys wrote: If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime peername from the CLI Without reload prune does not take effect in 1.4.x And after reload all registrations are lost. So basically Asterisk Realtime is big mess from our experience and is totally unusable. We ended making #exec based script which takes data from DB and forms static configuration on each reload. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpjūčio 20 d. 15:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mysql sip realtime Hi The column order in your mysql sip table is irrelevant (Example sip table here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) All generic parameters are still taken from sip.conf and you must set rtcachefriends=yes If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime peername from the CLI As stated previously, you should never have to reload the sip module once realtime is working properly Hope this all helps Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Strict Routing and canreinvite
Hello, I want to send Media outside Asterisk server, e.g. between peers. In CLI I see: . [Jun 8 13:13:58] VERBOSE[19112] logger.c: -- Native bridging SIP/5060-b7dc5218 and SIP/prov12-09ad3888 . [Jun 8 13:13:58] DEBUG[19112] chan_sip.c: Strict routing enforced for session 3ad367ee48778d2c523a60e62ae86...@85.113.41.129 And media still goes through Asterisk. Why is that? Why strict routing is enforced? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] error with dial timeout
Try this: Exten =_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1)) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: 2009 m. birželio 2 d. 11:07 To: asterisk-users@lists.digium.com Subject: [asterisk-users] error with dial timeout Hello, I am trying to do : Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Why? I forgot something ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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] Extract a MOS value from Asterisk CDR
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com
Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Formula here: http://www.nessoft.com/kb/50 has jitter in it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: Marc Leurent [mailto:lf...@leurent.eu]
Sent: 2009 m. balandžio 2 d. 13:56
To: asterisk-users@lists.digium.com
Cc: Mindaugas Kezys
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Check this:
http://www.voip-info.org/wiki/index.php?page=Call+Quality+Metrics
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Sorry for replying for the second time, but this issue is interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this: http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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[asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels
As Asterisk has inner problems and channels very often locks we have such
script to restart Asterisk each midnight.
We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions
because there are too much changes which would brake our system
(realtime/sip/iax2/cdr/etc/etc).
Script soft hangups all alive channels in dirty way then kills Asterisk and
starts it up.
Hope it will be useful to somebody.
Corrections/comments welcome.
#! /bin/sh
# Script to restart asterisk softly by Kolmisoft
# crontab
# 0 0 * * * /usr/local/mor/asterisk_nice_restart.sh
# tell Asterisk do not accept new calls
asterisk -rx 'stop gracefully' /dev/null
# read all channels
asterisk -rx 'core show channels verbose' | sed '1d' /tmp/f1
cat /tmp/f1 | awk '{split ($0,a, ); print a[11]}' /tmp/f2
# hangup all alive channels
for i in `cat /tmp/f2`; do
asterisk -rx soft hangup $i /dev/null
done
# let asterisk to stop by itself
sleep 5
# kill remainings
killall -9 safe_asterisk
killall -9 asterisk
# start fresh and ready to work!
/etc/init.d/asterisk start
# clean
rm -rf /tmp/f1
rm -rf /tmp/f2
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
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Re: [asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels
Any guidelines how to solve locked channels problems? E.g. to find out which part of the code has problems and causes locks. Upgrade to newer versions are not an option. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: 2009 m. kovo 19 d. 17:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels Mindaugas Kezys escribió: As Asterisk has inner problems and channels very often locks we have such script to restart Asterisk each midnight. snip That is the things we must help to solve for not having to do to something like this on asterisk servers. Fortunately I use 1.4.22 version which has proved to me to be quite stable, judging from this uptime: System uptime: 1 week, 2 days, 19 hours, 22 minutes, 53 seconds Last reload: 10 hours, 31 minutes, 33 seconds Upgrading to 1.4.23.1 resulted in random core dumps (suspecting attended transfers issue) but unfortunately I've had no time to debug it and make a good bug report. My case is a 24/7/365 non-stop call center, so I didn't have another choice but to rollback. I hope some of us just can help asterisk be better by trying to use the latest version at least on testing environments, to not having to maintain an internal version and cherrypicking patches that may or may not resolve the issues that we could experience. Just my 2 cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] Script to softly restart Asterisk each midnight to clean locked channels
Locked channel does not react to 'soft hangup' command.
That's why it is called - LOCKED.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Sent: 2009 m. kovo 19 d. 18:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Script to softly restart Asterisk each
midnight to clean locked channels
On Thu, 19 Mar 2009, Miguel Molina wrote:
Mindaugas Kezys escribi?:
As Asterisk has inner problems and channels very often locks we have
such script to restart Asterisk each midnight.
snip
Why restart Asterisk, free up the channel...
From cron, you can clear up any calls over say 3 hours:
/usr/sbin/asterisk -rx show channels concise|awk -F : '($11 10800)
{print /usr/sbin/asterisk -rx \soft hangup $1 \}'|sh
You don't necessarily have to keep restarting it at midnight.
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP
Enough research will tend to support your
conclusions. - Arthur Bloch
A conclusion is the place where you got
tired of thinking - Arthur Bloch
227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E
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Re: [asterisk-users] Dial() application 'g' option
How to determine which channel hung up first?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: 2009 m. vasario 22 d. 04:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial() application 'g' option
On Saturday, February 21, 2009, Philipp Kempgen wrote:
To be quite precise the documentation says
---cut---
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
---cut---
So I would not expect the g option to have any effect if the
*source* channel hangs up.
I guess you should do any kind of logging or post-hangup calculations
in the h extension.
Thanks. I did wonder about that but carried out some experiments that
suggested it didn't matter which channel hung up first. I have two SIP
geographical numbers with different providers and I tried ringing one
from the other and got the same result no matter which handset I hung
up first.
Unfortunately, by the time the call gets to the h extension, the
original dialled number in ${EXTEN} is changed to h - so I won't be
able to carry out the desired logging there. Also, I suspect that
${DIALEDTIME} and ${ANSWEREDTIME} might be lost. That said, I'm only
interested in recording the accumulated time for outgoing calls via
one SIP trunk, so if I can tie that down with a channel name...
Some further experimentation is in order!
Thanks again,
--
Geoff
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Re: [asterisk-users] VoIP Information in CDRs
I'm trying to find the answer to the same question:
4. Find out who hangedup an answered call.
HANGUPCAUSE = 16
DIALSTATUS = ANSWERER
In both cases, so these variables does not help.
Can anybody help with this issue? Should be pretty simple to detect which
part hanguped the call first.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Costa Tsaousis
Sent: 2009 m. vasario 21 d. 17:30
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VoIP Information in CDRs
Hi,
I am trying to find a way to add the following info in CDRs (with asterisk
1.4.23.1):
1. Codec used
2. RTP QoS statistics
3. RTP IP of remote host
4. For answered calls, the peer that requested to end the conversation
I have managed to get 1 and 2 for the caller, like that:
exten = h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}
Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(aud
ionativeformat)}/${SIPCHANINFO(t38passthrough)} QOS=${RTPAUDIOQOS})
The problems I have so far:
1. CODEC
Codec is reported only for A-Leg.
When transcoding asterisk logs the above line as: slin for read / slin for
write / the codec of A-Leg / 0 for t.38.
Is there a way to get the codec for both legs of a call?
2. RTP Qos is reported only for A-Leg.
Also, asterisk seems to ignore the RTP statistics reports by B-Leg after the
BYE:
-- Executing [...@core-dialplan:3] Hangup(SIP/401-08231540, ) in new
stack
== Spawn h extension (core-dialplan, h, 3) exited non-zero on
'SIP/401-08231540'
Scheduling destruction of SIP dialog
'0aa4f73f5c9715b7661b50080a669...@10.11.12.1' in 6656 ms (Method: INVITE)
set_destination: Parsing sip:4...@10.11.12.43:5060;transport=udp
sip:4...@10.11.12.43:5060;transport=udp for address/port to send to
set_destination: set destination to 10.11.12.43, port 5060
Reliably Transmitting (no NAT) to 10.11.12.43:5060:
BYE sip:4...@10.11.12.43:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport
From: Office Line 1 sip:4...@10.11.12.1
sip:4...@10.11.12.1;tag=as1d9352fe
To: sip:4...@10.11.12.43:5060;transport=udp
sip:4...@10.11.12.43:5060;transport=udp;tag=0009b7aa1aaa51eb2c767e13-7fb3b3
4a
Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1
CSeq: 103 BYE
User-Agent: home.tsaousis.gr
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (core-dialplan, 422, 1) exited non-zero on
'SIP/401-08231540'
box*CLI
--- SIP read from 10.11.12.43:50539 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport
From: Office Line 1 sip:4...@10.11.12.1
sip:4...@10.11.12.1;tag=as1d9352fe
To: sip:4...@10.11.12.43:5060;transport=udp
sip:4...@10.11.12.43:5060;transport=udp;tag=0009b7aa1aaa51eb2c767e13-7fb3b3
4a
Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1
Date: Sat, 21 Feb 2009 14:29:42 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0
RTP-RxStat: Dur=4,Pkt=180,Oct=28800,LatePkt=0,LostPkt=0,AvgJit=0
RTP-TxStat: Dur=4,Pkt=183,Oct=29280
These SIP messages are being exchanged after the dialplan has executed the h
extension.
Is there a way to have RTP statistics for both legs?
3. RTP IP is not reported anywhere.
The RIP= variable I have above, reports the SIP IP, and again only for
A-Leg.
Is it possible to find out the RTP (not SIP) IPs for both legs?
4. Find out who hangedup an answered call.
I have not found any way to determine the peer that requested to hangup the
call.
Is it possible to find who of the two legs requested the hangup?
Any help is appreciated.
Costa
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[asterisk-users] Caller Hangup detection
Hello, Is here any dial plan variable which could help me to identify that call was dropped (when still not connected) by caller? HANGUPCAUSE returns 0 DIALSTATUS returns NOANSWER How to identify such situation? Related question - how to know which end (caller or callee) ended the call first after call was answered? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] H323 stress test
Hello, We made small stress-test for H323. Test shows that H323 protocol is heavyweight compared with SIP. More details: http://wiki.kolmisoft.com/index.php/H323_pass-through_test Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Background stress test
Hello, We did small test with sipp to test Asterisk Background command capability. Our goal was 700 sim. calls on HP Proliant DL160 G5 E5405 1 x Quad Core Xeon 2Ghz 2 Gb RAM Asterisk 1.4.18.1 Centos 5.2 We reached more then 1000 when our network (100mbps) become a bottleneck. As we achieved our goal - no further testing was performed. As conclusion - we are very happy with Asterisk in this case. If somebody is interested - more details are here: http://wiki.kolmisoft.com/index.php/Asterisk_Background_performance_test Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Using Asterisk to measure call quality: Introducing Recqual
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: Looks very interesting. After reading all available info I have two questions before testing: 1. Who/what answers the calls at the other end? I guess real live traffic should be sent through this Asterisk server? 2. How many calls you had made to to diagnose your problems? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] H323 crashes Asterisk on high load
Hello, Asterisk 1.4.18.1 PWlib 1.10.0 Openh323 1.18.0 ../asterisk/channels/h323 compiled from source. Under high load H323 crashes and kills Asterisk, debug shows: (gdb) bt #0 0x007a2b18 in strcmp () from /lib/libc.so.6 #1 0x014478a1 in find_call_locked (call_reference=13, token=0xa1cc570 ip$81.192.72.46:7768/13) at chan_h323.c:1148 #2 0x01449f07 in cleanup_connection (call_reference=13, call_token=0xa1cc570 ip$81.192.72.46:7768/13) at chan_h323.c:2290 #3 0x0145a724 in MyH323EndPoint::OnConnectionCleared () from /usr/lib/asterisk/modules/chan_h323.so #4 0x00e604f1 in H323Connection::OnCleared () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #5 0x00e721d1 in H323EndPoint::CleanUpConnections () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #6 0x00e722fe in H323ConnectionsCleaner::Main () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #7 0x005fd6e5 in PThread::PX_ThreadStart () from /usr/local/lib/libpt_linux_x86_r.so.1.10.0 #8 0x0088446b in start_thread () from /lib/libpthread.so.0 #9 0x00804dbe in clone () from /lib/libc.so.6 Server 2x XEON quad core and 4g DDR crashes on 110-120 simm. H323 calls. Anybody experienced same situation? Maybe there is some fix? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] New release of billing and routing software MOR
Hello, We are glad to announce new release of our advanced billing and routing package for Asterisk - MOR v0.7 It is complete solution for VoIP billing and routing for advanced and start-up telecoms, carriers, voip calling card operators and ISPs. Demo available online, as LiveCD or as InstallCD. Contact us for more details. More info: http://www.kolmisoft.com What is new in this version: * Call Routing by priority (Manual LCR) * LCR/Tariff change based on call prefix * PBX Functions - small functions which extends functionality of MOR PRO * PDF UTF8 support * More statistical data * New permission system * Accountant role * CallerID Manipulation: * Localization/Provider Rules * CallerID change on Forward * SIP debug system * New payment gateways: LinkPoint and CyberPlat * Google Maps integration to show Active Calls on the map!!! * IVR system * Limit calls per provider/did/user/device basis * User/Device/DID import from files * Send invoices by email in batches * NO ANSWER/BUSY interpretation for providers * Currency engine rework - automatic update from web Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] strange h323 delay issue
Hi, Try downgrade to Asterisk 1.4.18.1. It works for us perfectly with H323. Following versions has nasty bugs, not actually related to H323, but who knows, maybe it will help to downgrade. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys Sent: 2008 m. spalio 18 d. 22:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] strange h323 delay issue Hello, I have a strange h323 issue. After executing command Dial(SIP/333-0d1dfe00, H323/[EMAIL PROTECTED]|5|tT) at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what asterisk (h323 channel) was doing for 40 sec??? What reasons could invoke this problem? I haven't any problems with SIP channels. My versions: asterisk-1.4.21.1 asterisk-addons-1.4.6 openh323_v1_18_0 pwlib_v1_10_0 My h323.conf configurations: [general] port = 1720 bindaddr = 192.168.1.165 tos=lowdelay disallow=all allow=g729 dtmfmode=rfc2833 gatekeeper = DISABLE AllowGKRouted = no AcceptAnonymous = no context=from-trunk [ccg] type=friend context=from-trunk host=192.168.1.163 port=1720 disallow=all ;allow=alaw ;allow=ulaw allow=g729 fastStart=yes h245Tunneling=yes A full log: [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/333-0d1d8fb0, H323/[EMAIL PROTECTED]|5|tT) in new stack [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: type=H323, format=8, [EMAIL PROTECTED] [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Extension: 361737052390920 Host: ccg [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Calling to [EMAIL PROTECTED] on H323/ccg-2 [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Requested transfer capability: 0x00 - SPEECH [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Placing outgoing call to [EMAIL PROTECTED]:1720, 101 [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper. [Oct 18 22:32:23] VERBOSE[18236] logger.c: Using 192.168.1.165 for outbound call [Oct 18 22:33:03] VERBOSE[18236] logger.c: == New H.323 Connection created. [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- root is calling host [EMAIL PROTECTED]:1720 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Call token is ip$localhost/6453 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Call reference is 6453 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- DTMF Payload is [pt=101] [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Called [EMAIL PROTECTED] [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting capabilities for connection ip$localhost/6453 [Oct 18 22:33:03] VERBOSE[18238] logger.c: Setting capabilities to 0x100 (g729) [Oct 18 22:33:03] VERBOSE[18238] logger.c: Capabilities in preference order is (g729) [Oct 18 22:33:03] VERBOSE[18238] logger.c: Allowed Codecs: [Oct 18 22:33:03] VERBOSE[18238] logger.c: Table: [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729A 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729 2 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/hookflash 3 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/RFC2833 4 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/dtmf 5 [Oct 18 22:33:03] VERBOSE[18238] logger.c: Set: [Oct 18 22:33:03] VERBOSE[18238] logger.c:0: [Oct 18 22:33:03] VERBOSE[18238] logger.c: 0: [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729A 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729 2 [Oct 18 22:33:03] VERBOSE[18238] logger.c: 1: [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/hookflash 3 [Oct 18 22:33:03] VERBOSE[18238] logger.c: 2: [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/RFC2833 4 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/dtmf 5 [Oct 18 22:33:03] VERBOSE[18238] logger.c: [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Capabilities for connection ip$localhost/6453 is set [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Created RTP channel [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting NAT on RTP to 0 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Sending SETUP message [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Transmitting RFC2833 on payload 101 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Started logical channel: sending G.729A [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- channelsOpen = 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c: External RTP
Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI
Congratulations with the release! I'm curious also about the statement on your page: Realtime Asterisk uses the MySQL relational database to access dialplan, extension and configuration data. This allows for dynamic additions and changes to users, extensions and dialplans without having to restart or reload the system. What version of Asterisk are you using? From my experience starting from 1.4.19 Asterisk Realtime is completely broken: 1. http://bugs.digium.com/view.php?id=12362 2. http://bugs.digium.com/view.php?id=12925 3. http://bugs.digium.com/view.php?id=12921 Also how do you go about changing details for device in DB and not using sip realtime prune PEER + 'sip reload'? Without that your changes to devices are not active. Good luck! Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Thursday, August 14, 2008 12:07 AM To: asterisk-users@lists.digium.com; Commercial and Business-Oriented Asterisk Discussion Subject: [asterisk-users] New GUI for Realtime Asterisk - RAGUI Our company, WebPoint IT Solutions has just released an open source (GPL V2 license), Ruby on Rails based gui manager for Realtime Asterisk called RAGUI. RAGUI is definitely a work in progress and has rough edges, but we expect to polish it up in the upcoming weeks and months. All comments, contributions, and criticisms are welcomed! Here are the links: Sourceforge: http://sourceforge.net/projects/ragui/ Website: http://www.ragui.net Enjoy! Mike Clark WebPoint IT Solutions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whitepaper: How and to whom sell VoIP
Hello, Based on our own and our clients' experience we compiled short manual: How and to whom sell VoIP Hope it can be useful to some of you also. You can download it from our site: http://www.kolmisoft.com Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP
Hi, Try to delete whole column 'md5secret' from DB peers table. Leave only 'secret'. And try then. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Walter Stanish Sent: Monday, July 21, 2008 8:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. It looks like Asterisk is unhappy with the SIP REGISTER request coming from your softphone for some reason. It's very strange that it's occurring for two different softphones though. Trun on SIP debugging by typing sip debug on your Asterisk console and then post up the 4 SIP messages invloved in the register transaction so we can take a look and spot why it could be getting rejected. Sure. Here's what happens when kphone starts up: == --- SIP read from 192.168.0.25:5060 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C CSeq: 35 REGISTER To: Walter sip:[EMAIL PROTECTED] Expires: 900 From: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: Walter sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER black*CLI - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.25 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 35 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED];tag=as59de1023 Call-ID: [EMAIL PROTECTED] CSeq: 35 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7864265a Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) == Kphone prompts for a password, then the following occurs. == --- SIP read from 192.168.0.25:5060 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C CSeq: 36 REGISTER To: Walter sip:[EMAIL PROTECTED] Authorization: Digest username=walter, realm=asterisk, nonce=7864265a, uri=sip:192.168.0.2, cnonce=abcdefghi, nc=0001, response=10a7024959390c04b4d09c708fac6130, opaque=, algorithm=MD5 Expires: 900 From: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: Walter sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.25 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 36 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED];tag=as59de1023 Call-ID: [EMAIL PROTECTED] CSeq: 36 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 [Jul 22 00:59:38] NOTICE[2414]: chan_sip.c:15049 handle_request_register: Registration from 'Walter sip:[EMAIL PROTECTED]' failed for '192.168.0.25' - Wrong password Scheduling destruction of SIP dialog '[EMAIL
[asterisk-users] MOR billing and routing 0.6 released
Hello, We are proudly to present new version of our billing and routing system MOR v0.6 More info: http://www.voip-info.org/wiki/view/MOR Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to pass variable between 2 Asterisk servers over IAX2
Hello, Anybody can advice how to pass variable between 2 Asterisk servers over IAX2? With SIP I can use SipAddHeader. How do to the same with IAX2? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to pass variable between 2 Asterisk servers over IAX2
Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, June 27, 2008 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pass variable between 2 Asterisk servers over IAX2 On Friday 27 June 2008 10:07:18 Mindaugas Kezys wrote: Anybody can advice how to pass variable between 2 Asterisk servers over IAX2? With SIP I can use SipAddHeader. How do to the same with IAX2? In 1.6, with IAXVAR(). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major problem with 1.4.21 asterisk
Same here. Some of our clients upgraded from 1.4.18.1 to 1.4.21. After some time CLI stops responding and no calls are possible. Killall -9 is the only way to solve (get out) of this situation till next time it hangs. Example CLI screenshot: http://193.138.191.205/packets/asterisk1.4.21_noresponse.jpg Back to 1.4.18.1 (1.4.19.x is even more broken: http://lists.digium.com/pipermail/asterisk-users/2008-April/209342.html). Regards, Mindaugas Kezys http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Wednesday, June 25, 2008 7:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Major problem with 1.4.21 asterisk Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime and OOH323
Not sure about OOH323, but H323 can: http://www.voip-info.org/wiki/view/Asterisk+RealTime+H323 Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: Saturday, June 21, 2008 10:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime and OOH323 Can Realtime be used with OOH323 ala sip_buddies? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Hello, Our company did 200+ installations around the globe and had no issues with stability with correct Asterisk version. We used most of 1.4. As far as I remember 1.4.16 has some nasty bugs along with 1.4.19.x (SIP + realtime). So current stable is 1.4.18.1 (for us). For load check: http://wiki.kolmisoft.com/index.php/How_fast_MOR_can_perform It shows how our billing application performs on top of Asterisk (2049 channels) and we can push it even further with some improvements. We DO NOT RESTART our Asterisk installations daily or weekly. They work for months. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Benoit Plessis Sent: Tuesday, May 06, 2008 2:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in Production ? Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). 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
[asterisk-users] RTCP stats
Hello, Is here an easy way to get RTCP Stats in channel variables after the call ends? Or source should be edited to accomplish this? I would like to know this before developing this feature. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX ___ -- 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] keep one line open
Check who is dialing this line by CallerID, if it is not your user - just drop the call. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gilbert saunders Sent: Thursday, April 17, 2008 8:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] keep one line open hi i have multiple lines going to my asterisk box etc 0282549087 , 028 3659874 , 0285469658 etc. is it possible to keep users from using the 0282549087 line always open that it only allows a certain user to make outgoing calls on it? _ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try http://us.rd.yahoo.com/evt=51733/*http:/mobile.yahoo.com/;_ylt=Ahu06i62sR8H DtDypao8Wcj9tAcJ%20 it now. ___ -- 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] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
AGX-Addons crashes Asterisk for us. Working solution (on 100+ servers we installed): - apt-get -y install g++ libtiff4 libtiff4-dev patch autoconf automake libtiff-tools cd /usr/src wget http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20080402.tar. gz tar xzvf spandsp-20080402.tar.gz cd /usr/src/spandsp-0.0.4 ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig cd /usr/src wget http://193.138.191.205/packets/fax_apps_asterisk14.tgz tar xzvf fax_apps_asterisk14.tgz cd /usr/src/fax_apps make make install Restart Asterisk. Voila! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Sent: Wednesday, April 16, 2008 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing No progress at all. Version from Debian/Lenny repository still crashes and I'm not able to compile AGX. It gives out a long list of error messages. Some unsatisfied dependencies...? I Can't experiment for a while after unwanted night-time visit of fire-fighters :-( I have to let everything dry and clean out of sand and drywall pieces :-( Martin - Original Message - From: Justin Newman mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: 11. dubna 2008 13:00 Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing Did this just start happening with the 1.4 tree? Have you made any progress on getting it resolved? Justin Newman Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: Let's be more specific here, folks: What version numbers? Asterisk, spandsp, agx-addons / rx-tx-fax? Asterisk: yesterday's 1.4 SVN SpanDSP: tried with pre 15, 16 and 18 AGX-Addons: tried with 1.4.5 and svn trunk rx/txfax: supplied by AGX Addons - although they seem to build the files and stick them into the modules directory, rather than adding to the apps directory and modifying the Makefile. i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 search=0xD6506D20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ -- 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] problem with Asterisk 1.4.19 - accountcode dissapearing
As far as I noticed - this issue is not 1.4.19 only. Same thing happens on
all Asterisk versions.
Set your own variable before transfer:
Exten = , Set(__MYACC=${CDR(accountcode)})
And use ${MYACC} in other (transfered) calls.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, April 15, 2008 3:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] problem with Asterisk 1.4.19 - accountcode
dissapearing
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS CORRECT AND IS
EQUALS TO THE ACCOUNTCODE SPECIFIED MUCH EARLIER IN THE DIALPLAN
exten = _X!.,n,Gotoif($[${i} = 1]?$[${PRIORITY}+2])
;DIAL ALL MAC PHONE ASSOCIATED WITH THIS EXTENSION SIMULATENOUSLY
exten =
_X!.,n,Dial(${mac_dial_string:0:$[${LEN(${mac_dial_string})}-20]}|${sip_phon
es_ring_time}) ;remove least 7 characters, thos
e are left there by the invalid last SQL fetch
exten = _X!.,n,Set(i=0)
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS EMPTY, and so
is this variable if I use it in any way.
When I dial an extension and it hits this diaplan, it works fine. But if I
dial an extension, answer and then transfer (using Polycom phones) to an
extension using this dialplan I lose the accountcode where specified in the
code. It's empty. How can ${CDR(accountcode)} lose it's value for no
reason in those two seemingly innocent diaplan lines?
Below is the CLI output if it's useful:
-- Executing [EMAIL PROTECTED]:22]
NoOp(SIP/0004f2134384-1-097fb4e8, 1234567890) in new stack ;THIS IS THE
ACCOUNTCODE
-- Executing [EMAIL PROTECTED]:23]
GotoIf(SIP/0004f2134384-1-097fb4e8, 0?25) in new stack
-- Executing [EMAIL PROTECTED]:24]
Dial(SIP/0004f2134384-1-097fb4e8, SIP/0004f2134384-3|8) in new stack
-- Called 0004f2134384-3
-- SIP/0004f2134384-3-099947b0 is ringing
== Spawn extension (generic-extensions-db, 705, 24) exited non-zero on
'SIP/0004f2134384-1-097fb4e8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back from
192.168.1.6
-- Nobody picked up in 8000 ms
-- Executing [EMAIL PROTECTED]:25]
Set(SIP/0004f212ae63-1-099700a8, i=0) in new stack
-- Executing [EMAIL PROTECTED]:26]
NoOp(SIP/0004f212ae63-1-099700a8, ) in new stack ;MISSING ACCOUNTCODE
IS HERE
Mick
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Re: [asterisk-users] *21*number # diverting
Google is your friend: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced VoIP Billing Solution From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gilbert saunders Sent: Wednesday, April 09, 2008 4:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] *21*number # diverting hi me again im new at asterisk and really need some good tutoring on asterisk and call forwarding i dont understand it at all pls help i have attached my extensions.conf file if someone would be so kind to look at it and tell me what code i must enter to make *21*number diverting and #21# undiverting possible __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk 1.4.19 crash with Realtime using SIP peers
Hello,
Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
gdb asterisk /tmp/coreXXX shows:
Program terminated with signal 11, Segmentation fault.
#0 0xb6148968 in find_peer (peer=0xb6042768 test, sin=0x0, realtime=1) at
chan_sip.c:2547
2547if (!(hp =
ast_gethostbyname(tmp-value, ahp)) || (memcmp(hp-h_addr, sin-sin_addr,
sizeof(hp-h_addr {
Sorry, I have no time to read manual how to correctly put this into bug
tracker.
Back to 1.4.18.1
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
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Re: [asterisk-users] rxfax crashes Asterisk (segmentation fault)
Hello, Rxfax from agx-ags-addons always crashes for us also. You can download apps we use from: http://193.138.191.205/packets/fax_apps_asterisk14.tgz Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced VoIP Billing From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mark morreny Sent: Friday, April 04, 2008 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rxfax crashes Asterisk (segmentation fault) Hi, I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk 1.4.18. Everytime rxfax executes, Asterisk crashes: -- Executing [EMAIL PROTECTED]:1] Set(Zap/2-1, FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif) in new stack -- Executing [EMAIL PROTECTED]:2] RxFAX(Zap/2-1, /var/spool/asterisk-fax/1207322398.0.tif) in new st ack [Apr 4 23:20:35] NOTICE[23925]: chan_iax2.c:6025 update_registry: Restricting registration for peer ' iaxmodem' to 60 seconds (requested 50) [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Pages transferred: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size: - 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image resolution- 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Transfer Rate: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Bad rows- 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Longest bad row run - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Compression typea st_speech_unregister [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size (bytes) - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = Segmentation fault Is rxfax supposed to be working? What could have caused this problem? Thanks, Mark ___ -- 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] How to obtain SIPCHANINFO variables within custom application?
Hello,
How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough
variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my custom
application, because there's no chan_sip.h and I can't include it into my
application (maybe there's other way?).
I can do like this:
exten = _X,1,Set(PEERIP=${SIPCHANINFO(peerip)})
exten = _X,2,custom_app
and read PEERIP with
pbx_builtin_getvar_helper, but that's not an option for me.
Any help?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
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Re: [asterisk-users] How to obtain SIPCHANINFO variables within custom application?
Thank you!
You saved my day!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Tuesday, March 25, 2008 5:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to obtain SIPCHANINFO variables
within custom application?
On Tuesday 25 March 2008 07:51:13 Mindaugas Kezys wrote:
How can I get peerip, recvip, from, uri, useragent, peername,
t38passthrough variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my
custom
application, because there's no chan_sip.h and I can't include it
into my
application (maybe there's other way?).
I can do like this:
exten = _X,1,Set(PEERIP=${SIPCHANINFO(peerip)})
exten = _X,2,custom_app
char buf[80];
pbx_substitute_variables_helper(chan, ${SIPCHANINFO(peerip)}, buf,
sizeof(buf));
BTW, this is exactly how res_config_curl works.
--
Tilghman
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Re: [asterisk-users] How is uniqueid computed
Hello, Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 ) If call is transfered or it is leg2 then: Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 + 1) This is from observations, i can be mistaken. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, March 18, 2008 6:12 PM To: asterisk-users Subject: [asterisk-users] How is uniqueid computed Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- 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] Redundant Voicemail
Hello, This can help: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ein Bielaczyc Sent: Monday, March 17, 2008 3:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Redundant Voicemail Forgive me if this has been covered before. I did search but I was unable to find a reference. I am curious to know more about the possibility of using SQL to store voicemail as well as having more than one voicemail system accessing a central SQL database. Any information would be appreciated. Thank you all, in advance. -- Ein Bielaczyc [EMAIL PROTECTED] NOTICE: This E-mail (including attachments) is covered by the Electronic Communications Privacy Act, 18 U.S.C.2510-2521, is confidential and may be legally privileged. If you are not the intended recipient, you are hereby notified that any retention, dissemination, distribution or copying of this communication is strictly prohibited. Please reply to the sender that you have received the message in error, then delete it. ___ -- 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] T.38 SIP Issues
Hello, Higher speeds then 9600kbps are not permited by patents. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Friday, March 14, 2008 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T.38 SIP Issues Has someone submitted a bugreport regarding enabling 9600kbps fax? I always wonder why it would never negociate 14400kbps... when it did work a single page on fine resolution would take 4 minutes. Thank you very much for that link. I knew there had to be more possible configurations for T.38. I will give it a try... but I think I can get away without patching chan_sip.c, no? that just seems to enable higher bitrates. And Linksys SPA2102 is one of the exact devices I have in my lab. On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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 ___ -- 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] T.38 SIP Issues
Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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] Silencing VoiceMail() app in * 1.4.10
Hello, Just find this file in /var/lib/asterisk/sounds and change it to anything you like. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Horwich IT Services (Godwin Stewart) Sent: Friday, March 07, 2008 10:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Silencing VoiceMail() app in * 1.4.10 Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten = 2,1,Playback(/media/asterisk/answerphone-en) exten = 2,n,VoiceMail(2000,s) exten = 2,n,Playback(/media/asterisk/thankyou-en) exten = 2,n,Hangup() The 's' option to VoiceMail() silences the prompt, leaves the beep just before going into 'record' mode, but also plays back auth-thankyou after the user hits the # key. How can I suppress playback of auth-thankyou at the end or get VoiceMail() to play back a different file? Thanks in advance, -- Godwin Stewart - Horwich IT services ___ -- 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] Silencing VoiceMail() app in * 1.4.10
Hello, Then you can change channel language in front of VoiceMail() app and in appropriate place put auth-thankyou file which is recorded/made by you. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Horwich IT Services (Godwin Stewart) Sent: Friday, March 07, 2008 1:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10 On Fri, 7 Mar 2008 12:10:37 +0200, Mindaugas Kezys [EMAIL PROTECTED] wrote: Just find this file in /var/lib/asterisk/sounds and change it to anything you like. But that will break other applications that use the auth-thankyou sound, Authenticate() for a start (which I use elsewhere in order to remote check the voicemailbox). -- Godwin Stewart - Horwich IT services ___ -- 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] Voted most stable and easy to use phone?
Linksys SPA942. Tried most of available phones on the market. These phones sits on companies tables for more then a year. No problem at all, easy to use, nice(!) to use. I recommend to everybody. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, February 22, 2008 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voted most stable and easy to use phone? On Thu, Feb 21, 2008 at 7:32 PM, arkda [EMAIL PROTECTED] wrote: I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are pretty straightforward to manage via TFTP, and work really well with Asterisk. I agree, we've had zero trouble with these. Easy to install and they just work. ___ -- 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] Best ATA. Period.
Linksys SPA 2102. No issues at all. Period. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, February 20, 2008 11:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Best ATA. Period. Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works? ___ -- 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] SIP GSM
Cyber-Telecom's CT-V372 is same box as PorTech MV-372 but with more advanced firmware. It supports more functions, such as SMS sending. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Willcox Sent: Wednesday, February 20, 2008 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP GSM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Tuesday, January 29, 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP GSM With that sort of set up, If for example i get a 8 channel GSM gateway and the X100P can i make more than 1 concurrent call though the gateway with the X100P or does it only support 1 call at a time? What im looking to do is get a multi channel GSM gateway, and have the ability to make more than 1 call at once through it. The PorTech MV-372 works nicely with asterisk and is multichannel (2, if that counts!) Cheers, Ben ___ -- 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] chan_h323 requirements
This can help (script for Debian): apt-get install flex bison #dirty hack to prevent error from missing file cd /usr/include/linux touch compiler.h #PWLIB cd /usr/src wget http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz tar zxvf pwlib-v1_10_0-src-tar.gz cd pwlib_v1_10_0/ ./configure make make install make opt PWLIBDIR=/usr/src/pwlib_v1_10_0 export PWLIBDIR #OpenH323 cd /usr/src wget http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz tar zxvf openh323-v1_18_0-src-tar.gz cd openh323_v1_18_0/ ./configure make make opt make install OPENH323DIR=/usr/src/openh323_v1_18_0/ export OPENH323DIR cd /usr/src/asterisk/channels/h323/ make make opt cd /usr/src/asterisk ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig #or similar way #cp /usr/local/lib/* /usr/lib Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, February 21, 2008 10:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_h323 requirements Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- 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 get a clean, basic configuration?
We do: in modules.conf: noload = pbx_ael.so noload = pbx_dundi.so noload = res_config_pgsql.so noload = res_smdi.so in extensions.conf delete every context [default], [demo], whatever in sip.conf, iax.conf delete all peer/users if any Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vincent Sent: Thursday, February 21, 2008 4:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to get a clean, basic configuration? Hello I'm using a standard Asterisk install with default settings, and when I run reload, I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to 'us' == Parsing '/etc/asterisk/features.conf': Found == Parsing '/etc/asterisk/adsi.conf': Found == Parsing '/etc/asterisk/dundi.conf': Found == Parsing '/etc/asterisk/extensions.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4094 pbx_load_module: Starting AEL load process. [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4101 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. etc. How can I go and trim things down? Thank you. ___ -- 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] Echo() app doesn't work
Hello, Seems you do not answer your channel before executing Echo(): -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- Try this dialplan: exten = _X.,1,Playback(demo-echotest) exten = _X.,2,Echo() exten = _X.,3,Hangup or exten = _X.,1,Answer exten = _X.,2,Echo() exten = _X.,3,Hangup Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yassen Damyanov Sent: Saturday, February 02, 2008 10:08 AM To: Asterisk Users Mailing List Subject: Re: [asterisk-users] Echo() app doesn't work --- Tzafrir Cohen [EMAIL PROTECTED] wrote: -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' On which platform is that? Echo is executed, and exists without an error. Tzafrir, thank you very much for responding! Logs look the same everywhere (on all 32-bit platforms where Echo() doesn't work) and on the 64-bit xubuntu (where it does). The log says it exited non-zero, which does not seem normal to me, but nevertheless the log has that on the only working setup already mentioned. I guess it is not the platform but maybe some kernel stuff that breaks the thing... Please anyone, any hint? Thanks in advance! I paste here my original message for reference (no broken lines this time): -Original Message-- Date:Fri, 1 Feb 2008 17:01:56 -0800 (PST) From: Yassen Damyanov [EMAIL PROTECTED] Add to Address BookAdd to Address Book Add Mobile Alert Subject: Echo() app doesn't work To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed. Details: A. Platforms: -- AsteriskNOW 0.6 beta 32bit, updated; -- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2 -- Debian Etch 32bit with custom compiled kernel (Timer frequency 1000 Hz and couple more tweaks) and latest stable asterisk (1.4.17) compiled from source -- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10) -- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages (1.4.10) Echo() works only on the 64-bit setup. Does not work for all other cases. The Playback() app works fine in *all* cases. (The microphone is tested and works fine, so it's not that simple!) For some of the setups I established two separate extensions and they could talk to each other (so important things work, yes). The logs show the same, that is, just what would be normal: -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- My extentions.conf: -cut here--- [globals] [general] [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [outgoing_calls] [incoming_calls] [internal] exten = 500,1,Verbose(1|Echo test application) exten = 500,n,Echo() exten = 500,n,Hangup() exten = 501,1,Verbose(1|Playback test application) exten = 501,n,Playback(vm-review) exten = 501,n,Wait(1) exten = 501,n,Hangup() [phones] include = internal -cut here--- My iax.conf: -cut here--- [general] bandwidth
Re: [asterisk-users] G729 version to be downloaded for my machines
Download for Pentium4 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, January 30, 2008 10:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G729 version to be downloaded for my machines Hi List; The output of cat /proc/cpuinfo giving a [Intel (R) Pentium (R) D] so what is the g729 version I have to download to work with my machine? Any help? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Asterisk's DANGEROUS Transfer CDR's
Hello, For such cases we usually suggest to put 2 boxes in your infrastructure: 1. Main billing gateway - where all PBX'es are connected (all client's remote PBX'es and your Local PBX) 2. Local PBX - where user's without PBX'es are connected Then user connects in following way: User - Local PBX - Main GTW - PSTN That way you will be save from transfer issue and all your clients will be able to transfer their calls on Local PBX. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: Wednesday, January 30, 2008 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's - Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 9:24:14 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's The provider can disable transfers (which is what we do), but why can a PBX not still allow it? Our PBX customers all can do transferring... but that's because billing isn't needed THERE. The billing, if any, is done on our end, or their providers end. This really seems like a very small and moot point that is being blown up. Depends how much it could cost you I guess :). If you're not supporting transfers it's a moot point if you are it's a bit more interesting. If the receptionist needs to transfer the call, then she should be able to do that within the confines of her PBX... the transfer of her call should NEVER go back out her PBX back to the supplier, for if it does, her PBX now loses control of that call. Our customer base is residential and small business. They don't want to either pay for or support another a PBX thats what they've come to us for in the first place a lot of the time. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] Enterprise or Fedora?
My suggestion - use the distro which you know best. We use Debian (200+ installations). It works stable for us because we know how to achieve it. Others use Fedora/Centos - because they are experts in these systems. Stability and performance of the system does not depend on the distro - only on person who built this system. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LWATCDR Sent: Friday, February 01, 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Enterprise or Fedora? We are using Fedora because that is what the company we got our system from recommended. If I was doing the system myself I would throw in my vote for CentOS. I am using it for a database server and I have had no problems with it at all. It is about as stable and secure of Linux distro as I have ever used. If you do go with them I suggest kicking the CentOS team a few dollars. They do dang good work. On Feb 1, 2008 9:16 AM, love U. all [EMAIL PROTECTED] wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference Express yourself instantly with MSN Messenger! MSN Messenger ___ -- 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 ___ -- 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] Realtime device update weirdness
Hello,
We use Asterisk Realtime for our billing software. 200+ installations of
Asterisk with Realtime, but I see this for the first time.
Asterisk 1.4.17, Addons 1.4.5, No patches, no NAT - just plain simple
installation.
With debug I can see:
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:138 realtime_mysql: MySQL
RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = '109' AND host =
'dynamic'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '0.0.0.0', port = '0',
regseconds = '0' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 0 rows on table: devices
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '0.0.0.0', port = '0',
regseconds = '0' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 0 rows on table: devices
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '213.164.10.178', port =
'60854', regseconds = '1201750701' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 1 rows on table: devices
Notice update: UPDATE devices SET ipaddr = '0.0.0.0', port = '0', regseconds =
'0' WHERE name = '109'
Correct behaviour is: UPDATE devices SET ipaddr = '213.164.10.178', port =
'60854', regseconds = '1201750701' WHERE name = '109'
Why update to 0.0.0.0 is executed? It makes devices unreachable. When device
reregisters - it becomes available for short time - then again - update to
0.0.0.0. Why it is happening?
For temporaly solution i had to patch res_config_mysql.c at line 342, added
such lines:
if ((!strcmp(newparam, ipaddr)) (!strcmp(buf, 0.0.0.0))){
ast_log(LOG_DEBUG,MySQL RealTime: Avoided to update %s to %s
!!!\n, newparam, buf);
ast_mutex_unlock(mysql_lock);
return -1;
}
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
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Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 12:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 05:48:01 pm Mindaugas Kezys wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, December 20, 2007 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. were you ever able to get a solution for this? i seem the same problem when storing my sip trunks in mysql, using 1.4.16.2 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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] Realtime logic in Asterisk 1.4.16.1
Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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] Realtime logic in Asterisk 1.4.16.1
Thank you! Will it come to 1.4.16.3 or 1.4.17? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Sunday, December 30, 2007 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 06:30:09 Mindaugas Kezys wrote: Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. Please update to the latest SVN 1.4 -- this should have already been fixed. -- Tilghman ___ --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] Active Calls
Happy Holidays! When call starts and your billing script/application starts – enter info about call in DB, like: datetime(call start), src, dst, RINGING, uniqueid When dialing: Dial(whatever|M(answer_mark_macro)) Macro: answer_mark_macro will put updated info to same row in DB: datetime(call answered), src, dst, ANSWERED, uniqueid When call ends your billing script/application should delete record from DB for this call. You can put TRANSFER info to DB also when transfer occurs. Also you can put any other info you find usefull about your call – codecs/phone model which is dialing and so on. This method lets you retrieve call status info from DB without using AMI – thus not bothering Asterisk at all. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR – Advanced Billing for Asterisk PBX From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdul Sent: Monday, December 24, 2007 12:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Active Calls Hi Friends, Happy New Year I was developing billing system for my end user customers. I need to get Asterisk Active calls in MySQL database with full status of call likem ringing, UP and runtime? i will be thank full for your help and suggestion. Thank You _ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try http://us.rd.yahoo.com/evt=51733/*http:/mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20 it now. ___ --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] ip phone suggestion for Asia?
Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. We use them (SPA942) in our company. Everybody's happy. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky Sent: Thursday, December 20, 2007 6:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ip phone suggestion for Asia? Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --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] Realtime logic in Asterisk 1.4.16.1
Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --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] Realtime logic in Asterisk 1.4.16.1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Thursday, December 20, 2007 12:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On 00:12, Thu 20 Dec 07, Mindaugas Kezys wrote: Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Have a look at http://downloads.digium.com/pub/security/AST-2007-027.pdf That's why it's not working anymore -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? --- Thank you for pointing this, but I red this doc many times. It does not help. I tried to put username/password for my device - but it still is looking for dynamic. Does it mean I can't have anything else in host field for device except dynamic? Also this PDF states: An attacker may impersonate any user using host-based authentication without a secret, simply by guessing the username of that user. AFAIK host-based authentication is done by IP address. Username and password are not present. Following this I see no logic in above statements: host-based authentication without a secret - host-based auth. is always WITHOUT secret, and simply by guessing the username of that user - again - host-based auth. is always WITHOUT username If device (peer/user) has username/password - that's not HOST-BASED authentication. Correct me if I'm wrong. Question follows - how can I have host-based authentication in Realtime in Asterisk 1.4.16.1?? Maybe tommorow we will see Asterisk 1.4.16.2? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --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] Realtime logic in Asterisk 1.4.16.1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, December 20, 2007 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. -- Tilghman -- Thank you for explanation, but problem is that only this first query is executed: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' [Dec 20 00:04:12] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:04:12] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) That's it. No more queries. End of call. Why? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Hello everybody, Since 1.4 release our company installed more then 200 Asterisk servers using Asterisk 1.4 version. At start we had several bugs with SIP channel and CDR handling but starting from 1.4.6 or something it works without problems. We are really happy with 1.4 and thank you for your great job! Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: Saturday, December 15, 2007 12:57 PM To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] Answer Machine/Fax/modem detection
Maybe this can help: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tong Sent: Sunday, December 02, 2007 7:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Answer Machine/Fax/modem detection Has anyone sucessfully implimented a fax or modem detection dial plan? I'm originating calls from asterisk using a list of numbers and dropping the destination into an IVR menu but need to do something different if a modem or fax answers. I tried to use the NVBackgroundDetect() application but i think that is for receiving faxes only. Any help would be appreciated. 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 ___ --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] Best Prepaid Application?
If you have any questions - there's forum on www.kolmisoft.com/mor to ask questions and get answers. Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Monday, November 26, 2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Prepaid Application? Thank you for your answer! I'm going to try it! Have a nice day Mindaugas Kezys a écrit : You can try MOR FREE - it has nice gui and is very fast. LiveCD is available: http://www.kolmisoft.com/mor/content/view/83/95/ It is covered in extensive manual: http://www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func, fileinfo/id,25/ And yes - it's FREE as name suggests. Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Friday, November 23, 2007 7:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best Prepaid Application? Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch 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 ___ --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 ___ --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] To DB or not to DB?
Pros: 1. No need to reload Asterisk when you change settings 2. Changes are applied instantly 3. Easy to manage dialplan/users/settings 4. With properly programmed GUI you can give users some self-help services 5. No noticable overhead - dual xeon + 2gb ram does 400 simm. calls 6. You can have your DB on other server, that let's you connect several Asterisk servers to one DB - unified configuration Cons: 1. None Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Wednesday, November 28, 2007 6:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] To DB or not to DB? I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time Asterisk?) and I'd like to know why if anyone cares to comment. Thanks Alan -- The way out is open! http://www.theopensourcerer.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 ___ --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] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Rename to codec_g729.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so Copy to /usr/lib/asterisk/modules chmod 777 codec_g729.so restart Asterisk show translations Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fernando Berretta Sent: Monday, November 26, 2007 6:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Dear Mindaugas, I've already download the folowing files for testing codec_g729-ast14-gcc4-glibc-athlon-sse.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-icc-glibc-x86_64-core2.so But... no one of them seems to be working ___ --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] Best Prepaid Application?
You can try MOR FREE - it has nice gui and is very fast. LiveCD is available: http://www.kolmisoft.com/mor/content/view/83/95/ It is covered in extensive manual: http://www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func, fileinfo/id,25/ And yes - it's FREE as name suggests. Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Friday, November 23, 2007 7:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best Prepaid Application? Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch 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 ___ --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] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
For testing purposes you can try one of these: http://kvin.lv/pub/Linux/Asterisk/ Mindaugas Kezys http://www.kolmisoft.com Advance Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fernando Berretta Sent: Friday, November 23, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Hi, I'm trying to install g729 codec in an Athlon 64 x2 Dual core processor 4000+ but.. all packages I've download haven't worked. Could someone please let me know what package should I download ? Best Regards, Fernando [EMAIL PROTECTED] modules]# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4222.52 processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4219.18 ___ --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] Dialing time-out
You can always press # at the end of your number to send it to Asterisk. Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Thursday, November 15, 2007 6:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dialing time-out Ok, probably a dumb question. I believe I already I know the answer, but thought I would get feedback from others. One of the issues with user devices at the end Asterisk is dialing time out. This is a parameter within each hardware device. So if I set it to 3 seconds it appears from the moment after going off hook any key press starts a timer allowing me 3 seconds to enter the next number before Asterisk times out and generically says I'm am sorry that is not a valid extension. Now this is ok, of sorts. The fault in this is when you dial a valid number you are stuck waiting 3 seconds for the system to out pulse and connect. This clearly separates Asterisk from the traditional TDM platform behavior where a time out can be REAL LONG allowed people to dial at a snail's rate without upsetting the phone system but then immediately out pulsing when a number match is met, regardless if the number match is a 4 digit extension or 7 digit phone number. Is this one of the reasons and purposes Asterisk has a real-time option? Thanks, Jim ___ --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] pip tones in Monitor or MixMonitor
If you want peep every 15s, you should do:
[some_context]
exten = _X.,1,Set(LIMIT_WARNING_FILE=beep)
exten =
_X.,2,Dial(Local/mixmoncontext/#{EXTEN}||L(3600:3600:15000)\n)
[mixmoncontext]
exten = _X.,1,MixMonitor...
In [some_context] use L option variables:
* LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the
caller.
* LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.
* LIMIT_TIMEOUT_FILE - File to play when time is up.
* LIMIT_CONNECT_FILE - File to play when call begins.
* LIMIT_WARNING_FILE - File to play as warning if 'y' is defined. If
LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce
(You have [XX minutes] YY seconds).
More details about L option for Dial cmd:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack
Sent: Thursday, November 15, 2007 12:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pip tones in Monitor or MixMonitor
I guess I didn't try this because Playback(beep) seems to me to playback the
beep once and not repeat ever 15 seconds as is needed for the pip tones.
Is this not true?
exten = _X.,1,Playback(beep)
exten = _X.,2,MixMonitor.
If you are starting the recording using some DTMF code sequence
described in features.conf make sure you use caller, callee or
both value to play sound to correct line end.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Plack Sent: Wednesday, November 14, 2007 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] pip tones in Monitor or MixMonitor
Is there a way to enable the pip tones (beep) indicating that a
call is being recorded?
I know that ChanSpy does beep (unless q option is chosen) once, but
not quite the same.
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Re: [asterisk-users] Problem with AGI Script
Make sure /usr/bin/perl can be reached.
Also try in your CLI:
agi debug
Same case happens when I do not have php-cli installed for php AGI scripts.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, November 14, 2007 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem with AGI Script
I have asterisk 1.2.18 running on a new system we just installed. Although
I've used AGIs many times in the past, I'm stumped on this one. It may just
be a simple issue that I need another eyeset to look at.
My AGI does the following:
#!/usr/bin/perl
#Load a few modules...
use Asterisk::AGI;
use DBI;
$AGI = new Asterisk::AGI;
#Grab input from Asterisk
my %input = $AGI-ReadParse();
#Some Debugging
$AGI-exec('SayDigits',$ARGV[0]);
exit;
All seems fine. If I run the script from the command line it works as
expected:
[EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
EXEC SayDigits 333
However, when actually running in practice I get:
-- Executing AGI(Zap/23-1, GetEmailfromDID.agi|5706016716) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
-- AGI Script GetEmailfromDID.agi completed, returning 0
extensions.conf
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
exten = s,3,rxfax(${FAXFILE})
exten = s,104,Set([EMAIL PROTECTED])
exten = s,105,Goto(3)
Any thoughts on why asterisk doesn't seem to be passing anything to the
script and the script doesn't seem to be passing anything back? When I call
I do not hear the digits read to me, instead I just get thrown to the next
object after the digit reading.
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Re: [asterisk-users] pip tones in Monitor or MixMonitor
exten = _X.,1,Playback(beep) exten = _X.,2,MixMonitor. If you are starting the recording using some DTMF code sequence described in features.conf make sure you use caller, callee or both value to play sound to correct line end. Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 14, 2007 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pip tones in Monitor or MixMonitor Is there a way to enable the pip tones (beep) indicating that a call is being recorded? I know that ChanSpy does beep (unless q option is chosen) once, but not quite the same. ___ --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] special kind of billing
You can try MOR: www.kolmisoft.com/mor It does what you need. It does it even in FREE version. PRO version costs _many_ times less then other not free solutions mentioned in this thread. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kate Kretz Sent: Wednesday, September 05, 2007 7:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] special kind of billing Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Cheers, Kate ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Multiple servers using realtime
Users register to (Open)SER which uses same DB as all Asterisk nodes. Asterisk Realtime engine lets change data in only one database to make changes global. (Open)SER does load-balancing and fail-over. You can even put second (Open)SER server in case first dies and use DNS SRV to make it active. Database (DB) can be on same machine, but it better should be dedicated to only DB to serve only queries from all nodes. Possible to use MySQL Replication and have same DB on all nodes, which will save some processing power. But it's harder to manage. There're tools, choice is yours how you use them. Regards/Pagarbiai, Mindaugas Kezys VoIP Billing Solutions http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Wednesday, August 29, 2007 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple servers using realtime On Tue, 2007-08-28 at 19:59 -0600, Edgar Guadamuz wrote: I have a confusion about using SER for balancing the load across the Asterisk boxes. The doubt is: once a user registers in a Asterisk box, all the calls from or to him are going to be done by the same Asterisk server or can a user make a call by one Asterisk server and then make another call by other Asterisk server? I think the user registers with the SER box. With loadbalancing an outgoing call can go through different Asterisk boxes: call #1 -- SER box #1 -- Asterisk box #1 -- destination call #2 -- SER box #1 -- Asterisk box #2 -- destination Regards, Patrick On 8/28/07, Dovid B [EMAIL PROTECTED] wrote: We have a similar set up. I would recommend also using SER and load balancing so you can load balance your calls out between your asterisk box's. ___ --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 ___ --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] Prepaid Billing: A2Billing, AstBill, ASTCC
You can try MOR FREE billing system for Asterisk. LiveCD can be downloaded from: http://www.kolmisoft.com/mor/index.php?option=com_contenttask=viewid=73 Regards/Pagarbiai, VoIP Billing Solutions Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Monday, August 27, 2007 1:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prepaid Billing: A2Billing, AstBill, ASTCC Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --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
