[asterisk-users] Why Local Channels are creating
Hi Does anyone know why Local Channels are creating in asterisk (1.6.1.1)?E.g. If I do call forward 4 channels and two threads are creating,it will delete after the call disconnected . In the 4 channel 2 of them of then are SIP channels and 2 of them are Local channel.Pleas tel me why this is using..? 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
[asterisk-users] Fail2Ban CSF
Hello list, anyone knows if fail2ban works together with CSF (http://www.configserver.com/cp/csf.html) ?? I use CSF for blocking port scanning and blocking of IP-adresses. I wonder if fail2ban will overwrite rules in iptables of CSF and vica versa. Kind regards, 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
[asterisk-users] DTMF not being heard correctly by far end conference system
Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some specific conference systems that they need. I am sure I saw this covered a few years ago but can't find it in the lists. The phones and the system are using rfc2833 and either alaw or ulaw, I have stayed away from in band dtmf, but may need to consider it. They also use *1 to turn on call recording and I am not sure how that will go with inband. Another 1.6 system has no problem with being detected and it uses SIP trunks from the same supplier as the customer. The first system is a 1.4.38 box, it has sip trunks as the primary outbound route, the secondary route is iax to another box then via analogue lines. Almost all the handsets are sip and a re a mix of polycom and yealink. The sip trunks routed out through the iax link via analogue lines seem to work okay too. I am wondering if the iax handling of dtmf matches whatever the far end is expecting a little better For now I have routed everything via the iax / analogue lines which may cause some problems in terms of line availability but gets past the issue. I am considering upgrading the box to 1.6 as the working one is 1.6 The other box is a digium AA50 appliance so I can't do much with it, other than find the right settings. I have on the first one relaxdtmf=yes - relates to old issues too as far as I can tell rfc2833compensate=yes - this only appears to matter for inbound I'm not sure these do anything useful From what I can tell it could be the toneduration, but don't know what it should be, and while technically its probably the IVR being fussy that doesn't help me and I want to see why one system works and one doesn't This is dtmf debug from an iax handset sending digit 4 [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-0639 to write format slin [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on 'SIP/xtreme-0639' [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237 [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5310 ast_channel_stop_silence_generator: Stopped silence generator on 'SIP/xtreme-0639' [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-0639 to write format alaw [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 1713844722 to 565606422 due to a source change [Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088) [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update [Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-0639 [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a source change [Jan 12 23:13:56] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF end on channel (IAX2/419-13088) [Jan 12 23:13:56] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update [Jan 12 23:13:56] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-0639 [Jan 12 23:13:56] DEBUG[8717]: res_features.c:1399 feature_interpret: Feature interpret: chan=IAX2/419-13088, peer=SIP/xtreme-0639, code=4, sense=1 I will get a sip dump but am remote for now and don't have sip access All pointers and knowledge appreciated Cheers Duncan -- _ -- 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] DTMF not being heard correctly by far end conference system
Am 12.01.2011 11:37, schrieb Duncan Turnbull: Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some specific conference systems that they need. I am sure I saw this covered a few years ago but can't find it in the lists. The phones and the system are using rfc2833 and either alaw or ulaw, I have stayed away from in band dtmf, but may need to consider it. They also use *1 to turn on call recording and I am not sure how that will go with inband. Another 1.6 system has no problem with being detected and it uses SIP trunks from the same supplier as the customer. The first system is a 1.4.38 box, it has sip trunks as the primary outbound route, the secondary route is iax to another box then via analogue lines. Almost all the handsets are sip and a re a mix of polycom and yealink. The sip trunks routed out through the iax link via analogue lines seem to work okay too. I am wondering if the iax handling of dtmf matches whatever the far end is expecting a little better For now I have routed everything via the iax / analogue lines which may cause some problems in terms of line availability but gets past the issue. I am considering upgrading the box to 1.6 as the working one is 1.6 The other box is a digium AA50 appliance so I can't do much with it, other than find the right settings. I have on the first one relaxdtmf=yes - relates to old issues too as far as I can tell rfc2833compensate=yes - this only appears to matter for inbound I'm not sure these do anything useful From what I can tell it could be the toneduration, but don't know what it should be, and while technically its probably the IVR being fussy that doesn't help me and I want to see why one system works and one doesn't This is dtmf debug from an iax handset sending digit 4 [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-0639 to write format slin [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on 'SIP/xtreme-0639' [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237 [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5310 ast_channel_stop_silence_generator: Stopped silence generator on 'SIP/xtreme-0639' [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-0639 to write format alaw [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 1713844722 to 565606422 due to a source change [Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088) [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update [Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-0639 [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a source change [Jan 12 23:13:56] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF end on channel (IAX2/419-13088) [Jan 12 23:13:56] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update [Jan 12 23:13:56] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-0639 [Jan 12 23:13:56] DEBUG[8717]: res_features.c:1399 feature_interpret: Feature interpret: chan=IAX2/419-13088, peer=SIP/xtreme-0639, code=4, sense=1 I will get a sip dump but am remote for now and don't have sip access All pointers and knowledge appreciated Cheers Duncan As far as I can remember you should take a look at the used codec and this here: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Some codecs do not feel happy with some seetings for dtmfmode. Perhaps you may comapre these on your 2 boxes. -- _ -- 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] Failed SIP registration kicks registered device off?
HI Ye, On Mon, Jan 10, 2011 at 10:04 AM, Ye Liu jaux...@gmail.com wrote: Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register extension 100 but with wrong password, so registration failed; 3. A then showed it was unregistered! Failed registration of device B shouldn't kick A off, I expect A stay online and work properly in this situation. Could anyone confirm this? Because my asterisk is modified, I'm not sure this behavior is in vanilla asterisk or it is caused by my own code. AFAIK, Asterisk does not support simultaneous registration from more than one device on the same extension. That is why you are seeing this behavior. As soon as B tries to register, the registration of A is 'overwritten'. If you need this behavior, you might want to try and look into a different UA Registrar like OpenSIPS, which supports this. Thank you! -- Ye Liu (AKA @jaux) -- James -- _ -- 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] Failed SIP registration kicks registered device off?
On 01/12/2011 10:07 AM, James Lamanna wrote: HI Ye, On Mon, Jan 10, 2011 at 10:04 AM, Ye Liujaux...@gmail.com wrote: Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register extension 100 but with wrong password, so registration failed; 3. A then showed it was unregistered! Failed registration of device B shouldn't kick A off, I expect A stay online and work properly in this situation. Could anyone confirm this? Because my asterisk is modified, I'm not sure this behavior is in vanilla asterisk or it is caused by my own code. AFAIK, Asterisk does not support simultaneous registration from more than one device on the same extension. That is why you are seeing this behavior. As soon as B tries to register, the registration of A is 'overwritten'. If you need this behavior, you might want to try and look into a different UA Registrar like OpenSIPS, which supports this. His point is valid though... A's registration should not have been overwritten until B *successfully* registered. A failed attempt to register should have no effect on the existing registration. -- 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 -- _ -- 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] Failed SIP registration kicks registered device off?
On Wed, Jan 12, 2011 at 10:13:22AM -0600, Kevin P. Fleming wrote: His point is valid though... A's registration should not have been overwritten until B *successfully* registered. A failed attempt to register should have no effect on the existing registration. Indeed, the avenue for a brute-force DoS (absent something like fail2ban) is fairly obvious. -- _ -- 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] Issue with Red Alarm with DAhDi
OpenVox A800P\ 8 port FXO *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Tue, 11 Jan 2011 17:09:51 -0600 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issue with Red Alarm with DAhDi On 1/11/11 2:33 PM, Edwin Quijada wrote: Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:46] WARNING[3411]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:47] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:52] WARNING[3412]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:53] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:58] WARNING[3413]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' I checked fisically the card and not red alarm in this. I am using Asterisk 1.4.38 and Dahdi 2.4.0 Any cluees ? TIA What card are you using for your DAHDI channels? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Issue with Red Alarm with DAhDi
On 1/12/11 10:29 AM, Edwin Quijada wrote: OpenVox A800P\ 8 port FXO I recommend contacting OpenVox for assistance with this. The Detected alarm on channel... message you see on the CLI is a direct result of a call from within the board driver. Date: Tue, 11 Jan 2011 17:09:51 -0600 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issue with Red Alarm with DAhDi On 1/11/11 2:33 PM, Edwin Quijada wrote: Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:46] WARNING[3411]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:47] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 snip I checked fisically the card and not red alarm in this. I am using Asterisk 1.4.38 and Dahdi 2.4.0 Any cluees ? TIA What card are you using for your DAHDI channels? Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] OpenVPN + SIP configuration?
On Tue, 11 Jan 2011 10:02:48 -0500, Mark Deneen mden...@gmail.com wrote: Using the shared secret will only allow a single point to point connection. That is, you have to use certificates if you want more than one client. Thanks for the tip. I was under the impression that the shared key is just the equivalent of the hashed password in /etc/shadow. Also, when running openvpn --genkey --secret static.key, I wasn't prompted for the hostname or IP address of the client, so I don't understand why using a shared key would limit connections only from a specific client. Or do you mean that once a client is connected, no other client can connect using the shared key? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On Tue, 11 Jan 2011 10:23:18 -0500, Bruce B bruceb...@gmail.com wrote: I have OpenVPN and Asterisk working nicely. However, I do use certificates. Though, it shouldn't matter. Can you explain what doesn't work for you? Is the connection not established or is the Asterisk and it's client not communicating? It's not working, because I'm stuck at what to put in the two configuration files, on either sides :-) Am I correct in understanding that we need three network addresses: - LAN were the server lives, eg. 192.168.0.0/24 - LAN where the client lives, eg. 192.168.1.0/24 - A third network number for the tunnel, eg. 192.168.2.0/24 ? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
Your network layout is correct. I am still unclear what is not working for you, but I guess you can't establish a connection yet. In the config file server.conf for the server side you will have parameter verb=3 which you can change to like 9 and see what the error message is upon connect. If you are using CentOS as client you can also check /var/log/messages on both client and server to see the error messages. You can also try ifconfig on the server side to make sure a Tun0 or a Tunx appears in your network address. -Bruce On Wed, Jan 12, 2011 at 12:14 PM, Gilles codecompl...@free.fr wrote: On Tue, 11 Jan 2011 10:23:18 -0500, Bruce B bruceb...@gmail.com wrote: I have OpenVPN and Asterisk working nicely. However, I do use certificates. Though, it shouldn't matter. Can you explain what doesn't work for you? Is the connection not established or is the Asterisk and it's client not communicating? It's not working, because I'm stuck at what to put in the two configuration files, on either sides :-) Am I correct in understanding that we need three network addresses: - LAN were the server lives, eg. 192.168.0.0/24 - LAN where the client lives, eg. 192.168.1.0/24 - A third network number for the tunnel, eg. 192.168.2.0/24 ? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI
Hi Everyone, I am looking for a paid version of a program that has proven to work with Outlook 2007 and Asterisk 1.6 on Windows Vista, XP, and maybe even Windows 7. Outcall is not the answer as it has lots of bugs and doesn't work. Something simple with very simple interface would be preferred. ***The program shall query Outlook contacts based on the Caller ID and open up the existing contact or open a New Contact form from Outlook. P.S. Outlook 2007 and Exchange Server 2003 are used. 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
Re: [asterisk-users] DTMF not being heard correctly by far end conference system
Hi Thorsten Thanks very much, at this point my preference is rfc2833 but I will try some other options. The system is generating audible tones (that I can hear), although I think the audio is generated by the last sip device in the network so if thats so I don't have any control of it. Probably then I have to go to inband to get some control back, I am not sure what I lose from this, or change upstream provider (although the current provider works from a different system) Cheers Duncan On 12/01/2011, at 11:42 PM, Thorsten Göllner wrote: As far as I can remember you should take a look at the used codec and this here: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Some codecs do not feel happy with some seetings for dtmfmode. Perhaps you may comapre these on your 2 boxes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with ZAP Channels
Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the soft hangup Zap/zapchannel in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes hidecallerid=no callwaiting = yes usecallingpres= yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=yes ;echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived loglevel=255 hanguponswitchpolarity=yes context=disc-from-trunk-ZAP001 pulsedial=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 busydetect=no busycount=5 callprogress=no cidsignalling=dtmf relaxdtmf=yes cidstart=polarity channel=1 Does anyone know what can i do to solve this problem? 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
Re: [asterisk-users] Problems with ZAP Channels
On Wed, 12 Jan 2011, Antonio Modesto wrote: Sometimes i am having problems with Zap channels on asterisk 1.2 Welcome fellow Luddite :) 1.2 is so old nobody cares. If you upgrade to a current release and still have problems, you will find a more receptive audience. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Queue periodic announce...
Is there a way to play a different message than the periodic announce after a certain time? I have been asked by a customer to do something like this: The user enters the queue. We play position and periodic announce every 60 seconds. If user has waited for more than 5 minutes then play a message with option to leave a voicemail This means that I would have to be able to play a different message after five minutes. I thought I could solve this by having the user leave the queue after 5 minutes, play the message with the option for voicemail and then reinserting them into the queue but I cannot find a way to put the user back in the same position they were when the timeout occurred. Any ideas on how to implement this? We are using Asterisk 1.6.2.15 with Queuemetrics. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] OpenVPN + SIP configuration?
On Wed, Jan 12, 2011 at 12:08 PM, Gilles codecompl...@free.fr wrote: On Tue, 11 Jan 2011 10:02:48 -0500, Mark Deneen mden...@gmail.com wrote: Using the shared secret will only allow a single point to point connection. That is, you have to use certificates if you want more than one client. Thanks for the tip. I was under the impression that the shared key is just the equivalent of the hashed password in /etc/shadow. Also, when running openvpn --genkey --secret static.key, I wasn't prompted for the hostname or IP address of the client, so I don't understand why using a shared key would limit connections only from a specific client. Or do you mean that once a client is connected, no other client can connect using the shared key? Thank you. From http://www.openvpn.net/index.php/open-source/documentation/howto.html#quick : Static Key disadvantages * Limited scalability -- one client, one server * Lack of perfect forward secrecy -- key compromise results in total disclosure of previous sessions * Secret key must exist in plaintext form on each VPN peer * Secret key must be exchanged using a pre-existing secure channel I honestly do not know what happens if you attempt to connect another client. It's either going to reject that client or disconnect the active one. -M -- _ -- 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] Queue periodic announce...
You'd have to use 2 queues. After 5 minutes, exit queue 1, enter queue 2 that has a different periodic announcement. Since everybody leaves queue 1 after 5 minutes, they will enter queue 2 in the same order as they left queue 1. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Wednesday, January 12, 2011 2:15 PM To: Asterisk Subject: [asterisk-users] Queue periodic announce... Is there a way to play a different message than the periodic announce after a certain time? I have been asked by a customer to do something like this: The user enters the queue. We play position and periodic announce every 60 seconds. If user has waited for more than 5 minutes then play a message with option to leave a voicemail This means that I would have to be able to play a different message after five minutes. I thought I could solve this by having the user leave the queue after 5 minutes, play the message with the option for voicemail and then reinserting them into the queue but I cannot find a way to put the user back in the same position they were when the timeout occurred. Any ideas on how to implement this? We are using Asterisk 1.6.2.15 with Queuemetrics. -- Telecomunicaciones Abiertas de M xico S.A. de C.V. Carlos Ch vez Prats Director de Tecnolog a +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] OpenVPN + SIP configuration?
On Wed, 2011-01-12 at 14:18 -0500, Mark Deneen wrote: Static Key disadvantages * Limited scalability -- one client, one server * Lack of perfect forward secrecy -- key compromise results in total disclosure of previous sessions * Secret key must exist in plaintext form on each VPN peer * Secret key must be exchanged using a pre-existing secure channel Yeah, that's all true. people claim that Openvpn is easier to configurate than ipsec, but the hardest part is: authentication/authorisation and routing. (which accidentally is with strongswan as easy/difficult as with openvpn ;-) When using self-signed certificates (both for the server and client) life isn't that hard: you can use step-by-step the info from the openvpn-web-site. Additional static key can be used to filter among valid certificate holders. Handy if you accept certificates from a (trusted) third party, but not all of them. (No, not Orwellians intended) hw -- _ -- 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] SetVar Warning
I had lines 3 and 4 and added line 1 and 2 to extensions.conf
exten = 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,2,Monitor(wav,${CALLFILENAME},m)
exten = 106,3,hint,SIP/106
exten = 106,4,Macro(stdexten,106,${HINT})
I received this warning:
WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for
extension (voicemenu-custom-4, 106, 1)
I'm running Asterisk/1.4.22.
Does anyone have any idea what I need to do to either make SetVar work or
replace it
with something else?
Thanks you,
Gary
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Re: [asterisk-users] SetVar Warning
On Wed, 12 Jan 2011, Gary Kuznitz wrote:
I had lines 3 and 4 and added line 1 and 2 to extensions.conf
exten = 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,2,Monitor(wav,${CALLFILENAME},m)
exten = 106,3,hint,SIP/106
exten = 106,4,Macro(stdexten,106,${HINT})
I received this warning:
WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for
extension (voicemenu-custom-4, 106, 1)
I'm running Asterisk/1.4.22.
Does anyone have any idea what I need to do to either make SetVar work
or replace it with something else?
I don't have a 1.4 system on hand, but 1.2 1.6 use set().
Also, just a suggestion to make your dialplan more maintainable, check out
the 'n' priority instead of explicitly numbered priorities.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Call hung up?
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT})
When I called x106 this was logged:
-- Executing [106@voicemenu-custom-4:1] Set(DAHDI/7-1,
CALLFILENAME=_xxx) in new stack
-- Executing [106@voicemenu-custom-4:2] Monitor(DAHDI/7-1, wav|_xxx-xxx-
|m) in new stack
== Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/7-1'
When I don't have the first two lines this is in the log:
-- Executing [106@voicemenu-custom-4:1] Macro(DAHDI/7-1,
stdexten|106|SIP/106) in new stack
-- Executing [s@macro-stdexten:1] Set(DAHDI/7-1, __DYNAMIC_FEATURES=)
in new stack
-- Executing [s@macro-stdexten:2] GotoIf(DAHDI/7-1, 0?5:3) in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [s@macro-stdexten:3] Dial(DAHDI/7-1, SIP/106|20|) in new
stack
What did I do wrong in adding the first two lines?
Thank you,
Gary
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Re: [asterisk-users] Call hung up?
On Wed, 12 Jan 2011, Gary Kuznitz wrote:
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT})
When I called x106 this was logged:
-- Executing [106@voicemenu-custom-4:1] Set(DAHDI/7-1,
CALLFILENAME=_xxx) in new stack
-- Executing [106@voicemenu-custom-4:2] Monitor(DAHDI/7-1, wav|_xxx-xxx-
|m) in new stack
== Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/7-1'
You are missing the priority on the 'macro' line.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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[asterisk-users] Polycom Blf / Directed Pickup
Would anyone happen to have some examples of polycom configs, specifically the 650 with sidecar for blf. I have the asterisk side all configured since I've set up blf with other types of phones, but I'm missing the polycom side. I've put together a mac-directory.xml, and the sidecar now lists numbers as speed dials but does not subscribe to blf. -- _ -- 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] Call hung up?
On Wed, 12 Jan 2011, Steve Edwards wrote:
On Wed, 12 Jan 2011, Gary Kuznitz wrote:
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT})
When I called x106 this was logged:
-- Executing [106@voicemenu-custom-4:1] Set(DAHDI/7-1,
CALLFILENAME=_xxx) in new stack
-- Executing [106@voicemenu-custom-4:2] Monitor(DAHDI/7-1,
wav|_xxx-xxx-
|m) in new stack
== Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/7-1'
You are missing the priority on the 'macro' line.
Also (at least in 1.2), the 'hint' line interferes with the interpretation
of 'n' on the 'macro' line. Try placing the 'hint' line first like:
[gary]
exten = 106,hint, SIP/106
exten = 106,1,
Set(CALLFILENAME=${TIMESTAMP}106_${CALLERID(num)})
exten = 106,n, Monitor(wav,${CALLFILENAME},m)
exten = 106,n, Macro(stdexten,106,${HINT})
The 'show dialplan' or 'dialplan show' (depending on version) command will
show you how Asterisk sees your dialplan which is not always like you
enter it in extensions.conf.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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