gmane.comp.telephony.pbx.asterisk.devel

[1.6.0] - chan_sip.c We could NOT get the channellock

Chris Maciejewski — 2008-08-28T08:58:49

Hi, Noticed when bridging two SIP calls in 1.6.0 I am getting the following error: ERROR[29457]: chan_sip.c:18560 handle_request_do: We could NOT get the channel lock for SIP/-0851d978! ERROR[29457]: chan_sip.c:18561 handle_request_do: SIP transaction failed: 1df6897d4c55dfb6041d762c20a4cbfb< at > However call is connected just fine and when looking into SIP packets flow it also look normal. Is this something to worry about? Regards, Chris _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Outgoing call file question

Steve Mathers — 2008-08-28T15:56:13

_______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

NOTICE[28890] chan_zap.c: PRI got event: HDLC Abort(6) on Primary D-channel of span 1

Sumit Gulati — 2008-08-28T11:57:27

Asterisk build-environment in Xen-DomU

Thorolf Godawa — 2008-08-27T17:43:13

Hi, I'm trying to migrate a Asterisk build-environment from a physical to a paravirtualized machine on Xen 3.0 - on this system Asterisk don't need to run at all, it is only for building RPMs. Host OS and guest OS both are CentOS 5.2. The build of Asterisk, Asterisk-addons and Zaptel works, but MeetMe and some other components are not compiled because Zaptel was not installed on the system. It seems not to be possible to install zaptel in a Xen-PV because of the lack of a RTC, so first try was to remove every thing RTC-related in zaptel-1.4.11 but I had no success with that. So I adapted the ztxen-1.4.2-patch to zaptel-1.4.11 and compiled it successfully. I installed asterisk-1.4.20.1, asterisk-addons-1.4.7 and zaptel-1.4.11 on my PV-system and adjusted the zaptel.conf that it only loads the ztxen. lsmod tells me that the module is loaded: Module Size Used by ztxen 7840 0 zaptel 190852 1 ztxen crc_ccitt 6337 1 zaptel But "make menuc

Memory Leaks (A janitor project?)

Steve Murphy — 2008-08-27T17:31:25

Off the wall idea - transmit wideband G722 or Speexover ISDN

Stephen Davies — 2008-08-27T06:24:13

cdr_mysql - do we really need "DESC cdr" every time?

Chris Maciejewski — 2008-08-26T19:28:40

Hello, I have just noticed cdr_mysql (addons 1.6) before inserting a CDR record into MySQL table, runs "DECS(ribe) cdr" _EVERY_ time. Would it not be better to load table structure (fields mappings) from cdr_mysql.conf and run ONLY "INSERT ...." statement? It seems to me we are causing unnecessary load on MySQL server and slowing things down. Regards, Chris _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

parse error in dahdi_cfg?

David A. Bandel — 2008-08-25T12:53:18

For the DAHDI developers: I know DAHDI isn't ready, but I checked-out dahdi and tools from trunk yesterday. I have a Wildcard w/ one FXS and one FXO module (dahdi_tools and dahdi_hardware report correctly). Just a couple of odd things: dahdi_cfg reports nothing with (and 0 errors) with echocanceller=mg2,1-2 however, if I put echocancellers=mg2,1-2 (note the 's' on echocanceller) I get 2 channels 'to be configured' and an error about the echocancellers argument. However, neither appear in asterisk after startup -- 'dahdi show channels' shows nothing configured and dahdi_tools shows the modules unconfigured. Odd that I _only_ get something if I include echocancellers= with the 's'. Parse error in the dahdi_cfg app? Ciao, David A. Bandel

zaptel/kernel/wct4xxp/base.c - FANCY_ALARM

Ming-Ching Tiew — 2008-08-24T22:34:09

Is there anyone who care to elaborate more about what is the significance of the macro FANCY_ALARM in the file zaptel/kernel/wct4xxp/base.c ? This is what is inside the file :- /* Define to get more attention-grabbing but slightly more I/O using alarm status */ #define FANCY_ALARM what is "more attention-grabbing" here ? Regards. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

SIP <=> IAX - and RTP

Darren Sessions — 2008-08-24T19:13:02

varget_helper and function

Kaloyan Kovachev — 2008-08-23T09:30:05

Hello, i have a custom app (with SVN-branch-1.4-r135597) which needs to read some channel variables and function results. The problem is that getvar_helper does not return any value, but always null. Is this a bug or i should use different builtin function for the function results like CDR(), CALLERID() and MUSICCLASS()? What about setting such values? Thank you. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Blind Transfer is not working in incoming calls

Max Alex — 2008-08-23T06:53:11

Voicemail has issues with DTMF

Max Alex — 2008-08-23T06:38:19

Losing SIP TLS connection on spurious INVITE

Bruce Atherton — 2008-08-22T19:19:52

I have an application that speaks SIP, and I am trying to certify that it works correctly with Asterisk 1.6 beta 9. In setting up a TLS connection, I send an invite and get an OK back. If the dialplan only has an Answer(), then all is good. But if I try to Dial() an extension, with or without Answer()ing first, I get an INVITE back to the application from * for some reason. This INVITE is not on port 5061, but on port 5060. This happens whether the extension being dialed is set up for TLS or not. As a result, a new dialog is created in my application. When the application attempts to say BYE using TLS on port 5060, it of course never gets anything back from *, and the original dialog is left hanging about. What is going on here? Why is there a new INVITE being sent to my application? If it is supposed to be there, why is it not on port 5061? Here is the SIP debug log from Asterisk. 192.168.168.83 is the * box, and 192.168.168.228 is where my application is running. The * box is the SIP proxy

apprendre astérisk

HONORA KOUADIO — 2008-08-22T17:09:01

frequent channel reset problem

Ming-Ching Tiew — 2008-08-22T15:04:01

Hi, I am stucked with a nasty PRI problem for 2 weeks now and will appreciate if I could get some help from here. The problem is that my zaptel.conf and zapata.conf have been working with one PRI line but it is not working with another PRI. I could easily blame it to the PRI line quality itself, but the fact is that the PRI line has been working fine with at least 2 PBXes for ages without experiencing channel reset or call drop problem. However when connected to asterisk using wct4xxp driver, it will get call drop randomly every now and then. From the asterisk console output, when calls are dropped, there is associated alarm detected. I would like to know the following: (1) After detecting an alarm, does the ISDN spec. specifies that all channels must be resetted or this is just an implementation choice which asterisk has made ? Should asterisk reset the line after detecting the alarm and caused all calls being drop? (2) I have a pri monitor ( ISDN tester) that sniffs the PR

H323 disconnect cause

2008-08-22T04:25:15

Hi, all We have the following scheme of the call: MVTS Softswitch -- H323 --> Asterisk -- SIP --> SIP Phone When SIP Phone is busy Asterisk gives release complete message with disconnect cause "Call rejected" (code 21) to MVTS Softswitch What we need is to get another disconnect cause ("User busy" with code 17) in this scheme of the call What should we patch in order to get code 17 on MVTS if the SIP Phone is busy? Asterisk 1.4.16.2 version OpenH323 1.18.0 version PWLib 1.10.0 version This software was compiled from source files downloaded from the official site www.digium.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-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

chan_sip.c:3707 sip_write: Asked to transmit frametype 64

Brad Finberg — 2008-08-22T01:30:51

app_fax.so and addons beta4

Germán Aracil Boned — 2008-08-21T21:50:55

Hi The module app_fax.so, don't exist with beta4 but exist with beta3 of asterisk addons.. Why ? I think this is a very good module for send and receive fax. regards

sip notify to reset cisco 79x1 phones

Pavel Jezek — 2008-08-20T22:38:19

Hi, if anyone have interest to implement into sip_notify mechanism to also send custom sip message bodies to phones... I'm attaching packet dump, that callmanager6 use to reset java based phones (79x1) currently, we are able to put sip header, using sip_notify.conf: Event: service-control Content-Lenght: 82 but callmanager also use message body, where is placed actual action to reset or restart phones... action=reset\n _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

56k SS7 using TE122.

Mark A Jenks — 2008-08-20T21:17:06