Re: [asterisk-users] Movistar sip Mexico
On 20/11/13 20:32 , Damian Gonzalez wrote: I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. I have had the same problem with a carrier, where some calls we receive from them have an image and an audio stream in the initial INVITE, even though the call is intended to use the audio stream. Responding back accepting T.38 will fail and *all* other options trying to reject the T.38 using known SIP supported methods will also fail. The *only* option is to just ignore the image stream, which is not allowed by the current set of SIP RFCs... Asterisk used to ignore the image stream, but since the 1.8(?) timeframe its behaviour has changed more towards standards compliance in this area. And now we're between a rock and a hard place. The only way out that I could find is to put something in front of Asterisk that just removes the image stream from initial INVITEs when received from the carrier. (OpenSIPS has this nice method called remove_stream() since a couple of versions) Complaining about this didn't help, Asterisk is not certified because Open Source, was basically their answer. -- Andreas Sikkema -- _ -- 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] 11.6 voicemail message cropped off?
Hey all I am running 11.6 and when a caller is sent to vociemail the greeting is cropped off and the beep occurs quickly. Incoming calls are g729 and this occurs where it is using the default greeting or a user provided greeting. I really want to go production with this are there any ideas what could cause an issue like this we have never seen it in 1.4 - 1.8 Bryant -- _ -- 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] 11.6 voicemail message cropped off?
Update When no greeting is recorded the default you have reached ext # greeting is cropped. When there is a greeting it is just ignored and not played at all. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Bryant Zimmerman brya...@zktech.com Sent: Saturday, November 23, 2013 8:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 11.6 voicemail message cropped off? Hey all I am running 11.6 and when a caller is sent to vociemail the greeting is cropped off and the beep occurs quickly. Incoming calls are g729 and this occurs where it is using the default greeting or a user provided greeting. I really want to go production with this are there any ideas what could cause an issue like this we have never seen it in 1.4 - 1.8 Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI Missing '/sys/bus/astribanks/drivers/xppdrv/sync'
Hello group, I am installing Asterisk on a new pc running Ubuntu 12.04.3 Server. I have dahdi-complete-2.7.0.1+2.7.0.1, and have tried 2.6.1 and 2.8.0-rc1 and 2.8.0-rc2. I am following along in the ASTERISK The Definitive Guide 4th Edition and on page 116 when I issue the command $ sudo /etc/init.d/dahdi start I get the following error: Loading DAHDI hardware modules: wctdm24xxp: done Running dahdi_cfg: done. Missing '/sys/bus/astribanks/drivers/xppdrv/sync' I found some steps at https://issues.asterisk.org/jira/browse/DAHLIN-325?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Here is the output of the commands listed there: $ sudo lspci -d d161:* -v 04:01.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) Subsystem: Digium, Inc. Wildcard TDM800P 8-port analog card Flags: bus master, medium devsel, latency 32, IRQ 16 I/O ports at d000 [size=256] Memory at f7c2 (32-bit, non-prefetchable) [size=1K] Expansion ROM at f7c0 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 Kernel driver in use: wctdm24xxp Kernel modules: wctdm24xxp $ sudo dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM800P name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM800P location=PCI Bus 04 Slot 02 basechan=1 totchans=8 irq=0 type=analog port=1,FXO port=2,FXO port=3,FXO port=4,FXO port=5,FXS port=6,FXS port=7,FXS port=8,FXS $ sudo dahdi_cfg -vvv DAHDI Tools Version - 2.7.0.1 DAHDI Version: 2.8.0-rc2 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08) 8 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 I have ran the command dahdi_genconf system and dahdi_genconf system modules trying to reload dahdi each time. Here is the output of my /etc/dahdi/system.conf file: # Autogenerated by /usr/sbin/dahdi_genconf on Sat Nov 23 14:42:33 2013 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/0 Wildcard TDM800P (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxsks=3 echocanceller=mg2,3 fxsks=4 echocanceller=mg2,4 fxoks=5 echocanceller=mg2,5 fxoks=6 echocanceller=mg2,6 fxoks=7 echocanceller=mg2,7 fxoks=8 echocanceller=mg2,8 # Global data loadzone = us defaultzone = us What any ideas? Thanks.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to answer a Panasonic PBX extension with Asterisk?
I'd like to have my Asterisk system pick up a certain extension on an existing Panasonic PBX when it rings. (It's connected to some proprietary Panasonic doorphones that I haven't replaced yet.) I connected that extension to an FXO port on a Digium AEX410 card, and set that channel to have the context doorphone. The problem is that the extension is never executed. With debugging on, I see: Nov 23 16:37:23] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring Begin on channel 12 [Nov 23 16:37:23] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (12) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN) [Nov 23 16:37:24] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring/Answered on channel 12 [Nov 23 16:37:24] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (12) - signaling (5) - event (ANALOG_EVENT_RINGOFFHOOK) [Nov 23 16:37:29] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring Begin on channel 12 [Nov 23 16:37:29] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (12) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN) [Nov 23 16:37:30] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring/Answered on channel 12 [Nov 23 16:37:30] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (12) - signaling (5) - event (ANALOG_EVENT_RINGOFFHOOK) so it seems to be detecting the ring, but that's it. What am I missing? -- Eric Cooper e c c @ c m u . e d u -- _ -- 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] Amazon, Asterisk and reliability beyond a hobby system?
On 11/22/2013 12:52 PM, Todd R. wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. I have only one small datapoint. I ran an EC2 microinstance with Asterisk and a dozen offboard users. The only problem I had was SIP wasn't dealing well with the Elastic IP one-to-one NAT that Amazon uses. I had the usual Asterisk/NAT issues of one-way audio. I eventually moved from Amazon to Linode to get away from the NAT issues. Once I did that, everything worked fine, but again it was only a dozen 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] Amazon, Asterisk and reliability beyond a hobby system?
Did you have the externalip setting in sip.conf set to the Elastic IP? Date: Sat, 23 Nov 2013 23:42:36 -0500 From: ja...@fivecats.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? On 11/22/2013 12:52 PM, Todd R. wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. I have only one small datapoint. I ran an EC2 microinstance with Asterisk and a dozen offboard users. The only problem I had was SIP wasn't dealing well with the Elastic IP one-to-one NAT that Amazon uses. I had the usual Asterisk/NAT issues of one-way audio. I eventually moved from Amazon to Linode to get away from the NAT issues. Once I did that, everything worked fine, but again it was only a dozen users. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
