[asterisk-users] DTMF logging
If I set the logging.conf to log DTMF it only seems to log dtmf messages that are bridged through the * server. If the call goes into a menu the DTMF dont get logged. Is the intended behavior? Scott England ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connection issues
We are almost consistently getting this error on users after they are connected anywhere from 2-10 minutes. The system immediately hangs up following this message. Jun 23 16:51:23 WARNING[17314]: chan_iax2.c:1717 attempt_transmit: Max retries exceeded to host xxx.xxx.xxx.xxx on IAX2/-14 (type = 6, subclass = 2, ts=479554, seqno=125) A google search did not reveal any useful information on this issue. Scott England ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hiss patch
In bug 0002863 a patch is mentioned that sends hiss every 20 seconds, does anyone know who wrote this or where it is available at? Scott England ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call disconnect
When connecting from providers (I have tried 3 now) in the UK and having the calls routed to my asterisk server in the US, I am suffering a call disconnect problem. The problem occurs whenever I record a call, either using record or sending the call to the voicemail application. This however does not occur when I route calls from US providers to the same Asterisk server.The calls are being disconnected after about 30-45 seconds of recording, and appear to be terminated normally. I can however listen to messages or stay on hold for 10 minutes or more. I have tried this on several other servers and get the same problem. Is there something I am missing in the configuration? Is is possibly due to there not being any audio being sent back to the PSTN gateway in the UK during a record function? Any help would be appreciated, I am really stumped on this one after many hours. Scott England ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK call disconnects during record
When receiving calls via IAX2 from providers in the UK I can only record a message 35 seconds long. However when I am receiving calls from other IAX providers I can record for as long as I desire. Has anyone have any ideas on this? Scott England ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Record Application
While trying to add the ability to record memos into my asterisk system I ran into a small issue with the Record app. If I specify a silence detection period the application makes the recording and hangs up,instead of continuing with the dial-plan as it should. Now if I dont specify a silence detection period an use only the # to end all recording sessions then everything works great, I make the recording, press # and the recording finishes and the system continues with the dial plan. With silence: exten = 205,2,Record(/tmp/asterisk-recording:gsm|5) Without Silence: exten = 205,2,Record(/tmp/asterisk-recording:gsm) I also receive this error message when I have the silence specified. NOTICE[475151]: Unable to find a path from ULAW to UNKN WARNING[475151]: Unable to restore read format on '[EMAIL PROTECTED]:4569]/2' This was happening on a remote * server without a zaptel interface. First I tried different codecs, that didnt work then I moved everything to a server that has a zaptel interface in it (Its the gateway for the PRI's that feed to the remote *) and the record function works as advertised, with or without a silence detection variable. Then to make sure it wasnt my config I got a voicepulse account and set it up to my remote * server and I have the same problem. So my conclusion here is that the Record function's silence detection only works on a machine with zaptel hardware. Is this new or am I just missing something in my configs. I have the dummy driver installed for timing is that possibly flawed? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP retries
Is there a way to increase the number of retries or the time to help with this? WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) Scott England ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP retries
Is there a way to increase the number of retries or the time to help with this? WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) Scott England ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF
I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the channel is in rfc2833 during the call via show channel. With SIP debug though I dont see any event for dtmf. I do see dtmf in IAX though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF
I dont expect to see an ascii code or such since the tones are in a rtp stream. But when I place the dtmf type to "info" in the sip.conf and make a call I see this under asterisk with sip debug on. DEBUG[122896]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 960, ms i s 140 I assume this is asterisk sending the dtmf tone, but if I switch to rfc2833 I dont see anything. What I am looking for is a way to verify * is sending the dtmf to the vocal server, since it does not see the dtmf even though the audio portion is in operation and I know dtmf works between the vocal server and a cisco AS5300. Scott England Sean P. Robertson wrote: I think that you are thinking of SIP INFO messages if you are expecting to see something in the SIP messaging. RFC2833 is sent as part of the RTP packets so you are not going to see a plain text 1,2,3,4,etc in a trace when using it. http://www.faqs.org/rfcs/rfc2833.html Sean - Original Message - From: "Scott England" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 5:58 AM Subject: [Asterisk-Users] DTMF I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the channel is in rfc2833 during the call via show channel. With SIP debug though I dont see any event for dtmf. I do see dtmf in IAX though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF
Vocal is a complex system and its very Cisco centric (since Cisco funds it I'm not suprised) but in all its been good. We went with it for the SIP support and the fact that our version is heavily modified to intgrate with our custom app. If * had solid SIP support 2 years ago things might have been diffrent. :) Scott costas wrote: I can't resist asking. What do you think of Vocal as compared to *? Anything Vocal has but missing in *? -- Original Message -- From: Scott England [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 17 Nov 2003 02:58:55 -0800 I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the channel is in rfc2833 during the call via show channel. With SIP debug though I dont see any event for dtmf. I do see dtmf in IAX though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and DTMF
Being relatively new to * I has what may be a simple question, I haven't been able to find it in the archives though, or at least been able to recognize it. I have a 400P the is acting as a pstn gateway. It then forwards via IAX2 to another * server at another site. The calls then get routed via callerid to a sip client with an exten statement in extensions.conf. However I cant seem to get DTMF to forward to the sip extension. With IAX debugging I can see the dtmf call at both iax points but I dont see it happen at the sip point. Do I have to do something different then a simple exten = 6000,1,SIP/[EMAIL PROTECTED] ? -- Scott England General Manager, ControlNet Inc. voice 408-850-4904 fax 408-866-4211 -- The information contained in this message may be privileged and confidential and protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by replying to the message and deleting it from your computer. Thank you. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
