Re: [asterisk-users] Choose IAX or SIP trunking?
On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote: 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? We use IAX trunks from our provider primarily as they are so much easier to configure and you only need one port open on your firewall/nat gateway. SIP needs hundreds, if not thousands of open ports IIUC. HTH Al ___ -- 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?
Alan Lord (News) wrote: On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote: 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? We use IAX trunks from our provider primarily as they are so much easier to configure and you only need one port open on your firewall/nat gateway. SIP needs hundreds, if not thousands of open ports IIUC. Yes, SIP requires very good, precise and complex stateful inspection from a firewall. You need 2 ports for each RTP stream, and these should be maintained by the firewall as media circuits are established and torn down. Do you know about any voice quality issues with IAX, especially with a big call volime? -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
[asterisk-users] Softphone in Web
Hello I am thinking to develop a softphone that is integrated into web.(in form of APPLET or some thing else) Ie a user with with just a PC with Net Browser(fire fox etc) Installed can make call.. Is there some thing developed before like this that is open source ?? -- Best Regards Shakeel Abbas ___ -- 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] Sending Dialled number down a sip channel to a PBX
Bumping this in the hope that it is seen by people who missed it before. Ishfaq Malik wrote: We have a customer who connects PBX boxes (Avaya etc.) to our asterisk server (1.4.17) as a SIP extension. This customer needs the dialled number sent to the PBX as well as number that the call is originating from so he can set up his own routing from his PBX box. I have tried setting both CALLERID(dnid) and CALLERID(rdnis) to the dialled number, though not at the same time but the customers PBX box does not pick up the dialled number setting. Has anyone got any experience in this? Thanks Ish -- 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
[asterisk-users] help on ${RTPAUDIOQOS}
Hi All,
While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in
my dialplan.
I had 2 sip extensions 555 and 666 and I called from 555 to 666, but
unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI.
Would you please let me know what is wrong with my dialplan and/or what else
should be done to get the value of ${RTPAUDIOQOS}?
Following is my dialplan context where my call landed
[incoming_vpbx]
exten = _x.,1,NoOp(A call has come)
exten = _x.,n,Noop(${RTPAUDIOQOS})
exten = _x.,n,Dial(SIP/666,30,m)
exten = _x.,n,Hangup()
exten = h,1,Noop(***${RTPAUDIOQOS})
And here is what appeared on CLI...
-- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7a80948, A call has
come) in new stack
-- Executing [...@incoming_vpbx:2] NoOp(SIP/555-b7a80948,
) in new stack
-- Executing [...@incoming_vpbx:3] Dial(SIP/555-b7a80948,
SIP/666,30,m) in new stack
== Using SIP RTP CoS mark 5
-- Called 666
-- Started music on hold, class 'default', on SIP/555-b7a80948
-- SIP/666-089cb090 is ringing
-- SIP/666-089cb090 answered SIP/555-b7a80948
-- Stopped music on hold on SIP/555-b7a80948
-- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090
-- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7a80948,
***) in new stack
Thanking you...
---Asterisk User
___
-- 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] Softphone in Web
ABBAS SHAKEEL a écrit : Hello Hi I am thinking to develop a softphone that is integrated into web.(in form of APPLET or some thing else) Ie a user with with just a PC with Net Browser(fire fox etc) Installed can make call.. Is there some thing developed before like this that is open source ?? Take a look at Mozphone -- Daniel ___ -- 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] Softphone in Web
Thanks. But Can i enhance it in such away that it can make calls to asterisk as part of a web application ?? user can call from webapplication i think mozphone is a plugin for mozilla... On Thu, Oct 1, 2009 at 2:00 PM, Administrator TOOTAI ad...@tootai.netwrote: ABBAS SHAKEEL a écrit : Hello Hi I am thinking to develop a softphone that is integrated into web.(in form of APPLET or some thing else) Ie a user with with just a PC with Net Browser(fire fox etc) Installed can make call.. Is there some thing developed before like this that is open source ?? Take a look at Mozphone -- Daniel ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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] got stuck at 150 calls, above that not working in stress test
On 1/10/09 5:56 PM, das sandesh wrote: Hi All, I have a problem, when I was doing a performance testing using an asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the other calls are giving busy, I tried to do ulimit related stuff, like increasing the soft and hard limits to 10 but no luck, Any ideas or views are really appreciated. Also I even changed the call limit to 500, but stills it can handle only 150 total. What do you mean it handles only 150? What happens when you get above that number? I regularly have more than that active on a machine. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Sending Dialled number down a sip channel to a PBX
On 1/10/09 9:24 PM, Ishfaq Malik wrote: Bumping this in the hope that it is seen by people who missed it before. Ishfaq Malik wrote: We have a customer who connects PBX boxes (Avaya etc.) to our asterisk server (1.4.17) as a SIP extension. This customer needs the dialled number sent to the PBX as well as number that the call is originating from so he can set up his own routing from his PBX box. I have tried setting both CALLERID(dnid) and CALLERID(rdnis) to the dialled number, though not at the same time but the customers PBX box does not pick up the dialled number setting. Has anyone got any experience in this? What are you dialing? If you have a remote box called [box_b] in sip.conf and you're dialing sip/1...@box_b, what do you get? Have you played around with all the options? Are you able to get a packet dump of how the packet should look? If so, you may be able to add a sip header if necessary. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] INVITE Sending Local IP
Hello. I set up an Asterisk box a couple days ago and was having problems with not being able to hear SIP clients. After some troubleshooting we have determined that hte INVITE is sending my local(192.168) IP. How would I get * to send the public IP instead of the local one? I have changed every IP/domain setting in sip.conf to reflect my public IP but it still doesnt want to work. Thanks to anyone hthat can help me. -Mike ___ -- 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] INVITE Sending Local IP
On 1 Oct 2009, at 10:43, Mike Bessette wrote: Hello. I set up an Asterisk box a couple days ago and was having problems with not being able to hear SIP clients. After some troubleshooting we have determined that hte INVITE is sending my local(192.168) IP. How would I get * to send the public IP instead of the local one? I have changed every IP/domain setting in sip.conf to reflect my public IP but it still doesnt want to work. Thanks to anyone hthat can help me. If you show us your config so we can see what is wrong... ___ -- 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] INVITE Sending Local IP
OK. Here is the relevant section of my sip.conf [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' ; if asterisk was compiled with OSP support. realm=windsorwebdynamic.com ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet domain=windsorwebdynamic.com; Set default domain for this host ; If configured, Asterisk will only allow ; INVITE and REFER to non-local domains ; Use sip show domains to list local domains domain=windsorwebdynamic.com ; Add domain and configure incoming context ; for external calls to this domain ;domain=1.2.3.4 ; Add IP address as local domain ; You can have several domain settings allowexternalinvites=yes; Disable INVITE and REFER to non-local domains ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) ;tos=184; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none ;maxexpiry=3600 ; Max length of incoming registration we allow ;defaultexpiry=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;checkmwi=10; Default time between mailbox checks for peers ;vmexten=voicemail ; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defaults to asterisk ;videosupport=yes ; Turn on support for SIP video ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; ;musicclass=default ; Sets the default music on hold class for all SIP calls ; This may also be set for individual users/peers ;language=en; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be rtptimeout) ;trustrpid = no ; If Remote-Party-ID should be trusted ;sendrpid = yes ; If Remote-Party-ID should be sent ;progressinband=never ; If we should generate in-band ringing always ; use 'never' to never use in-band signalling, even in cases ; where some buggy devices might not render it ; Valid values: yes, no, never Default: never ;useragent=Asterisk PBX ; Allows you to change the user agent string ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Note that promiscredir when redirects are made to the ; local system will cause loops since SIP is incapable ; of performing a hairpin call. ;usereqphone = no ; If
[asterisk-users] Friday Oct 2: Digium's new Speech Recognition for Asterisk
This week Steve Sokol stops by to describe and field questions about Digium's new affordable speech recognition solution. Later on in the call, we'll also be looking at iVoIP, clients and uses for mobile VoIP. Join us on IRC anytime #voip-users-conference During the conference, call via SIP g711 or wideband g722 - or Try the web page widget to call in wideband. The details on all the above are at http://VUC.me - pronounced Vee You See Me :) ___ -- 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] RTP Delayed during RTCP
Hello, Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? Can that be fixed? Thanks. ___ -- 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] portech MV-378 SIP GSM Gateway
Hi All I having an intermittent problem with the above mobile gateway and would appriciate some advice basically 1 in 10 calls fail at some point during the call, the duration of the calls ate completely different call progression Call comes in from Zap channel and dials a mobile number on the prtech gateway and it dials out on sip trunk 103, the call progresses ok and after a time the call goes silent without any warning any advice would be greatly appriciated Regards Robb ___ -- 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
[asterisk-users] Busy app timeout
Using 1.4 svn, I want to implent the busy application. With the following dialplan: [inboundqueue] exten = _X.,1,Answer() exten = _X.,n,Goto(dropcall,1) ... exten = dropcall,1,Busy(10) exten = dropcall,n,hangup() If I call any number in the inboundqueue, I get the following: [Oct 1 12:06:44] -- Executing [444...@isdnspan1:1] Answer(Zap/1-1, ) in new stack [Oct 1 12:06:44] -- Executing [444...@inboundqueue:2] Goto(Zap/1-1, 1?dropcall|1) in new stack [Oct 1 12:06:44] -- Goto (inboundqueue,dropcall,1) [Oct 1 12:06:44] -- Executing [dropc...@inboundqueue:1] Busy(Zap/1-1, 10) in new stack [Oct 1 12:06:44] == Spawn extension (inboundqueue, dropcall, 1) exited non-zero on 'Zap/1-1' why does the busy not wait for 10 seconds before dropping the zap channel ? show application Busy foxtrot*CLI -= Info about application 'Busy' =- [Synopsis] Indicate the Busy condition [Description] Busy([timeout]): This application will indicate the busy condition to the calling channel. If the optional timeout is specified, the calling channel will be hung up after the specified number of seconds. Otherwise, this application will wait until the calling channel hangs up. ___ -- 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] got stuck at 150 calls, above that not working in stress test
Hi Matt, When I get can more that 150 calls, i get a busy signal (Congestion) for the calls above 150 - says your call cannot be completed now, its allowing only 150 callsIs there any thing related to field descriptors from linux point of view that I need to increase inorder to increase the call capacity. Thanks Sandesh On Thu, Oct 1, 2009 at 4:19 AM, Matt Riddell li...@venturevoip.com wrote: On 1/10/09 5:56 PM, das sandesh wrote: Hi All, I have a problem, when I was doing a performance testing using an asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the other calls are giving busy, I tried to do ulimit related stuff, like increasing the soft and hard limits to 10 but no luck, Any ideas or views are really appreciated. Also I even changed the call limit to 500, but stills it can handle only 150 total. What do you mean it handles only 150? What happens when you get above that number? I regularly have more than that active on a machine. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] E1/T1 Tapping call recording in Asterisk - Testing needed
how can i used this patch with digium cards, i have digium card and also having some issue in recording , can you give me procedure for it? regards Dhaval On Thu, Oct 1, 2009 at 7:37 AM, Martin asteriskl...@callthem.info wrote: That's nice. At least now peopel that want to do call recording can do so without having to keep Asterisk in between the circuits. However all other applications like added voicemail, conferencing, followme etc ... still needs Asterisk in between unless they have a spare port on the PBX and do the routing... Martin On Wed, Sep 30, 2009 at 7:47 PM, Moises Silva moises.si...@gmail.com wrote: Is your code vendor locked to Sangoma ??? Hello Martin, not at all. The code is intended to be part of chan_dahdi Asterisk channel driver and as such any card capable of using the dahdi interface can benefit from it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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 ___ -- 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] INVITE Sending Local IP
Mike – It looks like you have externip set but no localnet setting. You need to set localnet for your internal networks so that Asterisk knows when to properly apply the externip setting. sl ___ -- 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] INVITE Sending Local IP
OK so basically just uncomment the the localnet settings hten? On Thu, Oct 1, 2009 at 8:15 AM, Scott L. Lykens slyk...@verimedservices.com wrote: Mike – It looks like you have externip set but no localnet setting. You need to set localnet for your internal networks so that Asterisk knows when to properly apply the externip setting. sl ___ -- 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] Sending Dialled number down a sip channel to a PBX
Ishfaq Malik wrote: Bumping this in the hope that it is seen by people who missed it before. Ishfaq Malik wrote: We have a customer who connects PBX boxes (Avaya etc.) to our asterisk server (1.4.17) as a SIP extension. This customer needs the dialled number sent to the PBX as well as number that the call is originating from so he can set up his own routing from his PBX box. I have tried setting both CALLERID(dnid) and CALLERID(rdnis) to the dialled number, though not at the same time but the customers PBX box does not pick up the dialled number setting. Has anyone got any experience in this? Thanks Ish I am no expert in this area, but my question would be 'Does sip support sending the called number on a trunk?'. Lyle ___ -- 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] RTP Delayed during RTCP
Cyprus VoIP wrote: Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? There is no such thing as an RTCP 'answer'. Can that be fixed? If it is a real problem, of course it can be fixed. The first step to doing so would be to actually provide enough details for others to try to understand the problem. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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
Re: [asterisk-users] Busy app timeout
Julian Lyndon-Smith wrote: Using 1.4 svn, I want to implent the busy application. With the following dialplan: [inboundqueue] exten = _X.,1,Answer() exten = _X.,n,Goto(dropcall,1) ... exten = dropcall,1,Busy(10) exten = dropcall,n,hangup() If I call any number in the inboundqueue, I get the following: [Oct 1 12:06:44] -- Executing [444...@isdnspan1:1] Answer(Zap/1-1, ) in new stack [Oct 1 12:06:44] -- Executing [444...@inboundqueue:2] Goto(Zap/1-1, 1?dropcall|1) in new stack [Oct 1 12:06:44] -- Goto (inboundqueue,dropcall,1) [Oct 1 12:06:44] -- Executing [dropc...@inboundqueue:1] Busy(Zap/1-1, 10) in new stack [Oct 1 12:06:44] == Spawn extension (inboundqueue, dropcall, 1) exited non-zero on 'Zap/1-1' why does the busy not wait for 10 seconds before dropping the zap channel ? Because (based on your log) the call came in over an ISDN circuit, and when you run the Busy application Asterisk sends a 'BUSY' indication to the calling switch, which then tears down the call. The timeout specified to Busy() is only relevant for channel types where the calling end will not drop the call on its own (like an analog channel). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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
[asterisk-users] Is there a way to get info who disconnected the call into CDR?
Hei! Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I'm setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present (completecaller, completeagent) Is there a way to get that info on the regular SS7 to SIP (and vica versa) calls? Best regards Rennes Neps ___ -- 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] INVITE Sending Local IP
Mike - Uncomment and set appropriately for your network. If you're using 192.168.1.0/24 as your internal network then that's what it should be set to. Be sure to include any private networks that may interact with the server over VPN or private circuits as well. Then be sure to reload or restart asterisk to make the changes effective. sl ___ -- 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] Is there a way to get info who disconnected thecall into CDR?
Found it, I use the g flag in Dial command, that helps :) Rennes From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rennes Neps Sent: 1. oktoober 2009. a. 16:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is there a way to get info who disconnected thecall into CDR? Hei! Here’s my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I’m setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present (completecaller, completeagent) Is there a way to get that info on the regular SS7 to SIP (and vica versa) calls? Best regards Rennes Neps No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.409 / Virus Database: 270.13.112/2391 - Release Date: 09/30/09 18:56:00 ___ -- 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] What are the reasons for VoIP echo?
I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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] INVITE Sending Local IP
Still no luck. I'm almost ready to start over with a fresh sip.conf and extensions.conf. Does anyone kno where I can find one without all the comments and other fluff? On Thu, Oct 1, 2009 at 9:22 AM, Scott L. Lykens slyk...@verimedservices.com wrote: Mike - Uncomment and set appropriately for your network. If you're using 192.168.1.0/24 as your internal network then that's what it should be set to. Be sure to include any private networks that may interact with the server over VPN or private circuits as well. Then be sure to reload or restart asterisk to make the changes effective. sl ___ -- 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] INVITE Sending Local IP
Mike – Your original post indicates the trouble is with audio. What kind of firewall are you passing through? If it’s PIX or ASA, I’ve found the most reliable route is to enable SIP inspect on the PIX/ASA and remove any externip/localnet configuration from Asterisk. This way the PIX/ASA is responsible for “fixing” SIP and making the media work. sl From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Bessette Sent: Thursday, October 01, 2009 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] INVITE Sending Local IP Still no luck. I'm almost ready to start over with a fresh sip.conf and extensions.conf. Does anyone kno where I can find one without all the comments and other fluff? On Thu, Oct 1, 2009 at 9:22 AM, Scott L. Lykens slyk...@verimedservices.com wrote: Mike - Uncomment and set appropriately for your network. If you're using 192.168.1.0/24 as your internal network then that's what it should be set to. Be sure to include any private networks that may interact with the server over VPN or private circuits as well. Then be sure to reload or restart asterisk to make the changes effective. sl ___ -- 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] INVITE Sending Local IP
Right now I have all firewalls and such turned off. When I have the firewall enabled, I use the one built in to the Tomato firmware on my Asus router. How could I determine if this is a PIX/ASA firewall? On Thu, Oct 1, 2009 at 10:33 AM, Scott L. Lykens slyk...@verimedservices.com wrote: Mike – Your original post indicates the trouble is with audio. What kind of firewall are you passing through? If it’s PIX or ASA, I’ve found the most reliable route is to enable SIP inspect on the PIX/ASA and remove any externip/localnet configuration from Asterisk. This way the PIX/ASA is responsible for “fixing” SIP and making the media work. sl *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike Bessette *Sent:* Thursday, October 01, 2009 10:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] INVITE Sending Local IP Still no luck. I'm almost ready to start over with a fresh sip.conf and extensions.conf. Does anyone kno where I can find one without all the comments and other fluff? On Thu, Oct 1, 2009 at 9:22 AM, Scott L. Lykens slyk...@verimedservices.com wrote: Mike - Uncomment and set appropriately for your network. If you're using 192.168.1.0/24 as your internal network then that's what it should be set to. Be sure to include any private networks that may interact with the server over VPN or private circuits as well. Then be sure to reload or restart asterisk to make the changes effective. sl ___ -- 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 ___ -- 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] E1/T1 Tapping call recording in Asterisk - Testing needed
On Thu, Oct 1, 2009 at 7:57 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: how can i used this patch with digium cards, i have digium card and also having some issue in recording , can you give me procedure for it? May be Martin can help with that, I don't know how to setup Digium boards in high impedance mode. It seems the feature may not be exported via configuration files yet, so changes to the driver may be needed? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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] E1/T1 Tapping call recording in Asterisk - Testing needed
Moises Silva wrote: May be Martin can help with that, I don't know how to setup Digium boards in high impedance mode. It seems the feature may not be exported via configuration files yet, so changes to the driver may be needed? That is correct, none of our drivers currently expose a method to put the framer interface into high-impedance mode. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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
Re: [asterisk-users] INVITE Sending Local IP
Mike - If your router/firewall does not have any kind of SIP protocol-specific support then you need to set up port forwarding on your router. Forward udp/5060 for signaling, and the matching udp ports as listed in your rtp.conf, to your Asterisk box. Keep the externip and localnet settings in place. Depending on how many ports are listed in rtp.conf you may choose to reduce this range but make sure the forwarding in your router matches the ports in rtp.conf. Read the below links to help determine an appropriate number of ports if you choose to change it. Some more information for you: http://www.voip-info.org/wiki/view/NAT+and+VOIP http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf PIX and ASA are specific models of firewalls from Cisco. They include a SIP protocol-specific fixup or inspect function that takes care of passing media through the firewall for you as long as you don't try to fix it yourself first. sl ___ -- 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] What are the reasons for VoIP echo?
At 07:10 AM 10/1/2009, you wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). Very similar to what I have. Also Flowroute for outgoing but others and a TDM400 for incoming. Since upgrading to 1.6.2 from 1.2.28 or so and figuring out DAAHDI and HPEC on the new version there have been no echo issues at all. Also cable modem but only the slow version. There is a Linksys router between the Asterisk box and the cable modem. Usually VOIP echo is the other end. There should be no way other than having the volume on your SIP phones way too high to get local echo. In the past it's been suggested on inexpensive phones with this problem to take apart the handset and make sure the path from the speaker to microphone is blocked or filled with sound insulation. I'm using Aastra 480i phones and that's never been a problem. Ira ___ -- 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] DTMF problems during a message play
I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the message to stop before the press is hear. I've tried all the suggestions found searching the wiki, so I ask here if there is something else I can try. The Vitelity trunk is set up as: dtmfmode=rfc2833 disallow=all allow=ulaw Thanks, Bart attachment: bhfisher.vcf___ -- 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] E1/T1 Tapping call recording in Asterisk - Testing needed
anyone can just grab the PEF framer datasheet and tweak the driver though... last I checked there's a whole section devoted to high impedance in the datasheet Martin On Thu, Oct 1, 2009 at 9:56 AM, Kevin P. Fleming kpflem...@digium.com wrote: Moises Silva wrote: May be Martin can help with that, I don't know how to setup Digium boards in high impedance mode. It seems the feature may not be exported via configuration files yet, so changes to the driver may be needed? That is correct, none of our drivers currently expose a method to put the framer interface into high-impedance mode. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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] portech MV-378 SIP GSM Gateway
Maybe the GSM carrier is disconnecting you ??? Just a wild guess. They sometimes do that if they have to free the channel ... for a better paying customer :) Martin On Thu, Oct 1, 2009 at 6:09 AM, robert boardman robert.board...@gmail.com wrote: Hi All I having an intermittent problem with the above mobile gateway and would appriciate some advice basically 1 in 10 calls fail at some point during the call, the duration of the calls ate completely different call progression Call comes in from Zap channel and dials a mobile number on the prtech gateway and it dials out on sip trunk 103, the call progresses ok and after a time the call goes silent without any warning any advice would be greatly appriciated Regards Robb ___ -- 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] QOS/DSCP for IAX?
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks ___ -- 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] QOS/DSCP for IAX?
I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks ___ -- 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] Voicemail - remove option to save in different folders
I checked the source for reading of configuration options but I didn't see anything in vm_execmain() This is the line of code that is bothering you cmd = get_folder2(chan, vm-savefolder, 1); On Mon, Sep 28, 2009 at 8:41 AM, Mike l...@virtutel.ca wrote: I am looking to configure the asterisk voicemail system to stop asking for the folder (work, personal, etc) in which to save messages when I do save them. Is there any configuration to do this? Mike ___ -- 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] QOS/DSCP for IAX?
Did you look at this wiki - http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 1:36 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] QOS/DSCP for IAX? I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks ___ -- 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] QOS/DSCP for IAX?
Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- 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] QOS/DSCP for IAX?
That link is great thanks. From what I read elsewhere, ToS is just the first 3 bits which should be honored by DSCP (first 5 bits)- even old equip should be DSCP compatible...or I need to do more reading :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- 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] QOS/DSCP for IAX?
Yea, kinda, sorta. The DSCP is six bits, which occupy six of the 8 bits in what is/was the type of service byte in an IP packet. Three of the 6 DSCP bits reside over the old precedence field and three reside over the old low delay, high throughput and high reliability fields (those three often referred to as TOS). The DSCP code points are designed to be backwards compatible with the PRECEDENCE portion of the old tos. The low delay, high throughput and high reliability bits have been redefined and no longer are backwards compatible. When doing my research I found some web sites displayed the tos byte in different bit-orders (cisco with precedence first, wikipedia with precedence last). It was confusing as heck. I also have some old equipment that does not understand DSCP/Diffserv. What I ended up doing was making asterisk and phones use the dscp code points and my old router software queue packets based on what it sees in the precedence field. Works like a charm. Good luck. -Dave Michelle Dupuis wrote: That link is great thanks. From what I read elsewhere, ToS is just the first 3 bits which should be honored by DSCP (first 5 bits)- even old equip should be DSCP compatible...or I need to do more reading :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- 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 ___ -- 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] got stuck at 150 calls, above that not working in stress test
On 2/10/09 12:41 AM, das sandesh wrote: Hi Matt, When I get can more that 150 calls, i get a busy signal (Congestion) for the calls above 150 - says your call cannot be completed now, its allowing only 150 callsIs there any thing related to field descriptors from linux point of view that I need to increase inorder to increase the call capacity. Is that coming from Asterisk? It seems strange that Asterisk would reject the call unless you have settings in asterisk.conf to do this. You've said you've already increased the file descriptor limits - did you do this in the console you were using to subsequently run Asterisk from? Do you get any errors in the Asterisk console? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] TDM410P - False Answer Supervision
Hello All, Can anyone help me with False Answer Supervision problem with TDM410P card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and everything works fine except the Answer supervision... When the call hits Asterisk it sends the call to one of the TDM410 card and the call is answered immediately while the call is still in progress... Here is the debug output: - [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing '3602045' [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3602045 [Oct 2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3602045w [Oct 2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9223421808-091b3f50 -- Hungup 'DAHDI/8-1' = The connect message is sent back immediately when DAHDI/8-1 answered SIP/9223421808-091b3f50 while the call is still in progress... If the call is hang up without answer the sender gets Normal Code 16 while it suppose to be Abandoned Call. The Polarity Reversal only works when call is ANSWERED... Here is the debug log: - [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing '3312808' [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3312808 [Oct 2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3312808w [Oct 2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9765782184-091b9678 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: Ignore switch to REVERSED Polarity on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: Ignoring Polarity switch to IDLE on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, aonp= 1, honp= 0, pdelay= 600, tv= 301564043 -- Hungup 'DAHDI/8-1' = Please help... Cheers, Nitesh ___ -- 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] TDM410P - False Answer Supervision
Assuming you're using POTS, you probably won't have much luck with this. If you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk won't process the line until you pick up and punch a dtmf key. If you are using E1 or PRI, there is more hope for you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Thursday, October 01, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TDM410P - False Answer Supervision Hello All, Can anyone help me with False Answer Supervision problem with TDM410P card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and everything works fine except the Answer supervision... When the call hits Asterisk it sends the call to one of the TDM410 card and the call is answered immediately while the call is still in progress... Here is the debug output: - [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing '3602045' [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3602045 [Oct 2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3602045w [Oct 2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9223421808-091b3f50 -- Hungup 'DAHDI/8-1' = The connect message is sent back immediately when DAHDI/8-1 answered SIP/9223421808-091b3f50 while the call is still in progress... If the call is hang up without answer the sender gets Normal Code 16 while it suppose to be Abandoned Call. The Polarity Reversal only works when call is ANSWERED... Here is the debug log: - [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing '3312808' [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3312808 [Oct 2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3312808w [Oct 2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9765782184-091b9678 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: Ignore switch to REVERSED Polarity on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: Ignoring Polarity switch to IDLE on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, aonp= 1, honp= 0, pdelay= 600, tv= 301564043 -- Hungup 'DAHDI/8-1' = Please help... Cheers, Nitesh ___ -- 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] What are the reasons for VoIP echo?
Ira writes: Very similar to what I have. Also Flowroute for outgoing but others and a TDM400 for incoming. Since upgrading to 1.6.2 from 1.2.28 or so and figuring out DAAHDI and HPEC on the new version there have been no echo issues at all. Also cable modem but only the slow version. There is a Linksys router between the Asterisk box and the cable modem. Usually VOIP echo is the other end. There should be no way other than having the volume on your SIP phones way too high to get local echo. In the past it's been suggested on inexpensive phones with this problem to take apart the handset and make sure the path from the speaker to microphone is blocked or filled with sound insulation. I'm using Aastra 480i phones and that's never been a problem. That's interesting. Our setup is almost identical to yours. We have 3 different SIP phones, one being an Aastra 480i. I have a Grandstream 4 Line one, and my wife has a Grandstream 2 line one. Both are relatively high-end phones (not inexpensive ones). The echo issues we've seen don't appear to be any different regardless of the phones in use. And strangely we never actually experience them on our end. The problem is always reported to us by someone calling in to us. This is what suggested to me to migrate my DID numbers over to them, rather than keeping them with Broadvoice. I think I might do that as a precaution anyway since its not likely to be a big problem for us to do it. The volume on our phones are not loud. I use a headset with my phone most of the time, which seems to give a better quality call to the other end, but there's not an issue with volume that could cause it. I am curious about the fact that you said after upgrading to 1.6.2, your problems went away. I didn't start with that version because it wasn't the current production version at the time. Do you think it would be beneficial to migrate to that version for me? Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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] Bringing people into a conference
The extension does exist, as the other caller is redirected to the room. Here's the relevant lines in extensions.conf: [dynamic-nway] exten = _XXX,1,Answer I've been trying to get this to work on and off for a while now, and it's time to get serious. If someone would like to get paid for getting this to work please contact me off-list (I also have a Google Wave invite if you're interested). The solution (and the steps we take) will of course be posted back here and you will also have my eternal gratitude. Thanks, Harley From: Matt Riddell li...@venturevoip.com To: asterisk-users@lists.digium.com Date: 23/09/2009 04:00 PM Subject: Re: [asterisk-users] Bringing people into a conference Sent by: asterisk-users-boun...@lists.digium.com On 23/09/09 5:07 PM, Harley Holcombe wrote: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. Is there an extension: dynamic-nway,282,1 Oh, and please refrain from using HTML emails to lists. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 Scanned by the Netbox from Netbox Blue (http://netboxblue.com/) Would you like total visibility and control over use of Web 2.0 applications such as media streaming, gaming and instant messaging at your company? If so, see: http://netboxblue.com/products/advancedidsips Scanned by the Netbox from Netbox Blue (http://netboxblue.com/) ___ -- 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] What are the reasons for VoIP echo?
At 02:53 PM 10/1/2009, you wrote: I am curious about the fact that you said after upgrading to 1.6.2, your problems went away. I didn't start with that version because it wasn't the current production version at the time. Do you think it would be beneficial to migrate to that version for me? I would expect not. I originally installed from an all in one disk and then removed most everything but asterisk. I always had issues with channels getting stuck and needing to reboot occasionally. I had tried 1.4 a few time and never got past 6 calls before it segfaulted so I stayed with 1.2, always kept up to date. When I built my new machine using an Atom and CentOS 5 I decided to use the 1.6.2 beta because I beta test a lot of software and I'm comfortable with the risks. It was probably silly, but all the issues I had before with stuck channels and other nonsensical problems are now a thing of the past. Echo has rarely been a problem for us once I got the echo cancellation working properly on the incoming only POTS line. I don't recall it happening on outgoing calls which are now either Flowroute for domestic or CallWithUs for international. Occasionally I'll get problems to New Zealand or Australia, but calling back usually fixes it. No clue why, but at 1.5 cents/minute I can put up with it. Ira ___ -- 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] TDM410P - False Answer Supervision
Danny, Thanks for your reply... Yes these are POTS line and I am not calling myself... Any other suggestions? Cheers, Nitesh Danny Nicholas wrote: Assuming you're using POTS, you probably won't have much luck with this. If you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk won't process the line until you pick up and punch a dtmf key. If you are using E1 or PRI, there is more hope for you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Thursday, October 01, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TDM410P - False Answer Supervision Hello All, Can anyone help me with False Answer Supervision problem with TDM410P card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and everything works fine except the Answer supervision... When the call hits Asterisk it sends the call to one of the TDM410 card and the call is answered immediately while the call is still in progress... Here is the debug output: - [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing '3602045' [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3602045 [Oct 2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3602045w [Oct 2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9223421808-091b3f50 -- Hungup 'DAHDI/8-1' = The connect message is sent back immediately when DAHDI/8-1 answered SIP/9223421808-091b3f50 while the call is still in progress... If the call is hang up without answer the sender gets Normal Code 16 while it suppose to be Abandoned Call. The Polarity Reversal only works when call is ANSWERED... Here is the debug log: - [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing '3312808' [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3312808 [Oct 2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3312808w [Oct 2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9765782184-091b9678 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: Ignore switch to REVERSED Polarity on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: Ignoring Polarity switch to IDLE on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, aonp= 1, honp= 0, pdelay= 600, tv= 301564043 -- Hungup 'DAHDI/8-1' = Please help... Cheers, Nitesh ___ -- 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 ___ -- 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] IAX2 Call rejected, CallToken Support required
Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the calltokenignore list or setting user priv requirecalltoken=no I have seen posts that suggest using: calltokenoptional = 0.0.0.0/0.0.0.0 or calltokenignore=xxx.xxx.xxx.xxx Using the above cause asterisk not to display the error but nothing occurs in the CLI. If I enable debug I see the following with the option calltokenoptional = 0.0.0.0/0.0.0.0 in iax2.conf in the general section. On the sending Server Asterisk 1.2.x Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 01471 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 138954087 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 01471 [192.168.42.251:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00010ms SCall: 03923 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 182789945 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 03923 [192.168.42.251:4569] Tx-Frame Retry[000] -- OSeqno: 091 ISeqno: 076 Type: VOICE Subclass: 136 Timestamp: 1048584ms SCall: 1 DCall: 2 [192.168.22.251:4569] Oct 2 10:05:41 NOTICE[32273]: chan_iax2.c:2880 auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.42.250:4569-4 is circuit-busy Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP On the receiving Server Asteirsk 1.4.x Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00657 DCall: 2 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 152361611 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 2 DCall: 00657 [192.168.25.250:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04012ms SCall: 2 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 I would really appreciate it if someone was able to give me an answer to this problem or at least point me in the right direction. Regards David. ___ -- 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] What are the reasons for VoIP echo?
if a user calling you hears echo of himself then it's the fault of your sip device/sip phone. The manufacturer must be using a cheap or an open source echo canceller ... try getting a different sip device made by some 'normal' company like polycom or linksys/cisco Martin On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham my...@techsol.org wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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] TDM410P - False Answer Supervision
Are you in US ? do you have the proper keywords in zapata.conf/chan_dahdi.conf like callprogress=yes etc ? Martin On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha nit...@vipernetworks.com wrote: Danny, Thanks for your reply... Yes these are POTS line and I am not calling myself... Any other suggestions? Cheers, Nitesh Danny Nicholas wrote: Assuming you're using POTS, you probably won't have much luck with this. If you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk won't process the line until you pick up and punch a dtmf key. If you are using E1 or PRI, there is more hope for you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Thursday, October 01, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TDM410P - False Answer Supervision Hello All, Can anyone help me with False Answer Supervision problem with TDM410P card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and everything works fine except the Answer supervision... When the call hits Asterisk it sends the call to one of the TDM410 card and the call is answered immediately while the call is still in progress... Here is the debug output: - [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing '3602045' [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3602045 [Oct 2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3602045w [Oct 2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9223421808-091b3f50 -- Hungup 'DAHDI/8-1' = The connect message is sent back immediately when DAHDI/8-1 answered SIP/9223421808-091b3f50 while the call is still in progress... If the call is hang up without answer the sender gets Normal Code 16 while it suppose to be Abandoned Call. The Polarity Reversal only works when call is ANSWERED... Here is the debug log: - [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing '3312808' [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3312808 [Oct 2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3312808w [Oct 2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9765782184-091b9678 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: Ignore switch to REVERSED Polarity on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: Ignoring Polarity switch to IDLE on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, aonp= 1, honp= 0, pdelay= 600, tv= 301564043 -- Hungup 'DAHDI/8-1' = Please help... Cheers, Nitesh ___ -- 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 ___ -- 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] What are the reasons for VoIP echo?
I'm quite new to all this but I was under the impression that most electrically induced echo was at the physical interface to the PSTN. If one is using SIP trunking, I would think this would point to a carrier issue. We also hit an interesting problem with echo today but I don't think this is the issue Myles is having. We installed fairly high end phones with full duplex speakerphones. Callers are having a bad problem with echo when the users use the speakerphone. Because it is full duplex rather than half, if the speakerphone volume and speakerphone mike volume are turned up, the callers are indeed hearing themselves by virtue of the higher quality full duplex! On Thu, 2009-10-01 at 19:36 -0500, Martin wrote: if a user calling you hears echo of himself then it's the fault of your sip device/sip phone. The manufacturer must be using a cheap or an open source echo canceller ... try getting a different sip device made by some 'normal' company like polycom or linksys/cisco Martin On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham my...@techsol.org wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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:
Re: [asterisk-users] What are the reasons for VoIP echo?
Are you saying there are half duplex phones out there with half duplex speakerphones ? All analog phones are full duplex ... Anyways the echo can be created by the analog phone even when it's connected to the sip ata or even the sip phone ... then you usually have acoustic echo which goes from speaker to microphone of the handset ... that should be cancelled by the sip phone/device... or someone out there will hear echo Martin On Thu, Oct 1, 2009 at 7:57 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: I'm quite new to all this but I was under the impression that most electrically induced echo was at the physical interface to the PSTN. If one is using SIP trunking, I would think this would point to a carrier issue. We also hit an interesting problem with echo today but I don't think this is the issue Myles is having. We installed fairly high end phones with full duplex speakerphones. Callers are having a bad problem with echo when the users use the speakerphone. Because it is full duplex rather than half, if the speakerphone volume and speakerphone mike volume are turned up, the callers are indeed hearing themselves by virtue of the higher quality full duplex! On Thu, 2009-10-01 at 19:36 -0500, Martin wrote: if a user calling you hears echo of himself then it's the fault of your sip device/sip phone. The manufacturer must be using a cheap or an open source echo canceller ... try getting a different sip device made by some 'normal' company like polycom or linksys/cisco Martin On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham my...@techsol.org wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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:
Re: [asterisk-users] What are the reasons for VoIP echo?
Indeed there are! - John On Thu, 2009-10-01 at 20:18 -0500, Martin wrote: Are you saying there are half duplex phones out there with half duplex speakerphones ? All analog phones are full duplex ... Anyways the echo can be created by the analog phone even when it's connected to the sip ata or even the sip phone ... then you usually have acoustic echo which goes from speaker to microphone of the handset ... that should be cancelled by the sip phone/device... or someone out there will hear echo Martin On Thu, Oct 1, 2009 at 7:57 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: I'm quite new to all this but I was under the impression that most electrically induced echo was at the physical interface to the PSTN. If one is using SIP trunking, I would think this would point to a carrier issue. We also hit an interesting problem with echo today but I don't think this is the issue Myles is having. We installed fairly high end phones with full duplex speakerphones. Callers are having a bad problem with echo when the users use the speakerphone. Because it is full duplex rather than half, if the speakerphone volume and speakerphone mike volume are turned up, the callers are indeed hearing themselves by virtue of the higher quality full duplex! On Thu, 2009-10-01 at 19:36 -0500, Martin wrote: if a user calling you hears echo of himself then it's the fault of your sip device/sip phone. The manufacturer must be using a cheap or an open source echo canceller ... try getting a different sip device made by some 'normal' company like polycom or linksys/cisco Martin On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham my...@techsol.org wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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
Re: [asterisk-users] IAX2 Call rejected, CallToken Support required
I had a problem between my 1.6.0 server and a 1.4 server trying to call through iax and I just put requirecalltoken=no in the stanza and that fixed the problem. Klaverstyn, David C david.klavers...@intergraph.com wrote: Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the calltokenignore list or setting user priv requirecalltoken=no I have seen posts that suggest using: calltokenoptional = 0.0.0.0/0.0.0.0 or calltokenignore=xxx.xxx.xxx.xxx Using the above cause asterisk not to display the error but nothing occurs in the CLI. If I enable debug I see the following with the option calltokenoptional = 0.0.0.0/0.0.0.0 in iax2.conf in the general section. On the sending Server Asterisk 1.2.x Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 01471 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 138954087 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 01471 [192.168.42.251:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00010ms SCall: 03923 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 182789945 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 03923 [192.168.42.251:4569] Tx-Frame Retry[000] -- OSeqno: 091 ISeqno: 076 Type: VOICE Subclass: 136 Timestamp: 1048584ms SCall: 1 DCall: 2 [192.168.22.251:4569] Oct 2 10:05:41 NOTICE[32273]: chan_iax2.c:2880 auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.42.250:4569-4 is circuit-busy Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP On the receiving Server Asteirsk 1.4.x Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00657 DCall: 2 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 152361611 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 2 DCall: 00657 [192.168.25.250:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04012ms SCall: 2 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 I would really appreciate it if someone was able to give me an answer to this problem or at least point me in the right direction. Regards David. Alternatives: ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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] TDM410P - False Answer Supervision
Thanks Martin, Well the Asterisk is in Fiji and we have check with the Telco on Reverse Polarity and they said it is setup... Here is my chan_dahdi.conf:- #include dahdi-channels.conf [channels] language=en context=incoming signalling=fxs_ks busydetect=yes callprogress=yes usecallerid=yes ;hanguponpolarityswitch=yes answeronpolarityswitch=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=2.0 txgain=3.0 group=1 callgroup=1 pickupgroup=1 channel = 1-4 channel = 5-8 Cheers, Nitesh Martin wrote: Are you in US ? do you have the proper keywords in zapata.conf/chan_dahdi.conf like callprogress=yes etc ? Martin On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha nit...@vipernetworks.com wrote: Danny, Thanks for your reply... Yes these are POTS line and I am not calling myself... Any other suggestions? Cheers, Nitesh Danny Nicholas wrote: Assuming you're using POTS, you probably won't have much luck with this. If you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk won't process the line until you pick up and punch a dtmf key. If you are using E1 or PRI, there is more hope for you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Thursday, October 01, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TDM410P - False Answer Supervision Hello All, Can anyone help me with False Answer Supervision problem with TDM410P card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and everything works fine except the Answer supervision... When the call hits Asterisk it sends the call to one of the TDM410 card and the call is answered immediately while the call is still in progress... Here is the debug output: - [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing '3602045' [Oct 2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3602045 [Oct 2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3602045w [Oct 2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9223421808-091b3f50 -- Hungup 'DAHDI/8-1' = The connect message is sent back immediately when DAHDI/8-1 answered SIP/9223421808-091b3f50 while the call is still in progress... If the call is hang up without answer the sender gets Normal Code 16 while it suppose to be Abandoned Call. The Polarity Reversal only works when call is ANSWERED... Here is the debug log: - [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing '3312808' [Oct 2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring dialing... -- Called G2/3312808 [Oct 2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent deferred digit string: T3312808w [Oct 2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done dialing, but waiting for progress detection before doing more... -- DAHDI/8-1 answered SIP/9765782184-091b9678 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: Ignore switch to REVERSED Polarity on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: Ignoring Polarity switch to IDLE on channel 8, state 6 [Oct 2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, aonp= 1, honp= 0, pdelay= 600, tv= 301564043 -- Hungup 'DAHDI/8-1' = Please help... Cheers, Nitesh ___ -- 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 ___ -- 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] IAX2 Call rejected, CallToken Support required
I tried your recommendation. I don't get an error with that but the call is cancelled with a debug of: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00011ms SCall: 00269 DCall: 3 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 182763616 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 3 DCall: 00269 [192.168.25.250:4569] Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04011ms SCall: 3 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 -Original Message- From: cov...@ccs.covici.com [mailto:cov...@ccs.covici.com] Sent: Friday, 2 October 2009 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Klaverstyn, David C Subject: Re: [asterisk-users] IAX2 Call rejected, CallToken Support required I had a problem between my 1.6.0 server and a 1.4 server trying to call through iax and I just put requirecalltoken=no in the stanza and that fixed the problem. Klaverstyn, David C david.klavers...@intergraph.com wrote: Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the calltokenignore list or setting user priv requirecalltoken=no I have seen posts that suggest using: calltokenoptional = 0.0.0.0/0.0.0.0 or calltokenignore=xxx.xxx.xxx.xxx Using the above cause asterisk not to display the error but nothing occurs in the CLI. If I enable debug I see the following with the option calltokenoptional = 0.0.0.0/0.0.0.0 in iax2.conf in the general section. On the sending Server Asterisk 1.2.x Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 01471 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 138954087 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 01471 [192.168.42.251:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00010ms SCall: 03923 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 182789945 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 03923 [192.168.42.251:4569] Tx-Frame Retry[000] -- OSeqno: 091 ISeqno: 076 Type: VOICE Subclass: 136 Timestamp: 1048584ms SCall: 1 DCall: 2 [192.168.22.251:4569] Oct 2 10:05:41 NOTICE[32273]: chan_iax2.c:2880 auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.42.250:4569-4 is circuit-busy Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP On the receiving Server Asteirsk 1.4.x Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00657 DCall: 2 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 152361611 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 2 DCall: 00657 [192.168.25.250:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04012ms SCall: 2 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 I would really appreciate it if someone was able to give me an answer to this problem or at least point me in the right direction. Regards David. Alternatives: ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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:
