Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
Yes, I am getting the exact same messages on my console if I do not answer the call as my first priority in the dialplan. On Jan 6, 2005, at 4:11 PM, John Middleton wrote: They are set to No HMMM I get message Ring/off-hook in strange state 6 on channel1 in the log Anyone seen this before On Thu, 6 Jan 2005 23:07:39 +, Phil Quinney <[EMAIL PROTECTED]> wrote: They should be set to no - sorry, I should have been clearer. Busydetect and callprogress cause my X100P's to not answer calls if I have them enabled. Phil. On 6 Jan 2005, at 22:56, John Middleton wrote: what should those two settings say? should i set them to yes, or take the lines out? OK: Do you have these in zapata.conf? busydetect=no callprogress=no They won't work in the UK. ___ 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 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
Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
Exactly. I know this is not preferred, but it is the only way I can get around the problem currently (strange state message). I am sure the call progress and busy detect are both no in my conf. I am looking for a "More Correct" answer to this as well. On Jan 6, 2005, at 4:34 PM, John Middleton wrote: What do you mean, as your first priority, you mean exten => s,1,Answer? On Thu, 6 Jan 2005 16:21:35 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote: Yes, I am getting the exact same messages on my console if I do not answer the call as my first priority in the dialplan. On Jan 6, 2005, at 4:11 PM, John Middleton wrote: They are set to No HMMM I get message Ring/off-hook in strange state 6 on channel1 in the log Anyone seen this before On Thu, 6 Jan 2005 23:07:39 +, Phil Quinney <[EMAIL PROTECTED]> wrote: They should be set to no - sorry, I should have been clearer. Busydetect and callprogress cause my X100P's to not answer calls if I have them enabled. Phil. On 6 Jan 2005, at 22:56, John Middleton wrote: what should those two settings say? should i set them to yes, or take the lines out? OK: Do you have these in zapata.conf? busydetect=no callprogress=no They won't work in the UK. ___ 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 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
Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
Here is the pertinent section of my extensions.conf: [from-pots] exten => s,1,Answer exten => s,2,Dial(sip/2001&sip/2002&sip/2003,20,t) exten => s,3,Voicemail(u2000) exten => s,103,Voicemail(b2000) exten => s,104,Hangup exten => a,1,VoicemailMain exten => a,2,Hangup exten => t,1,Hangup exten => h,1,Hangup On Jan 6, 2005, at 4:53 PM, John Middleton wrote: Sorry, maybe I didn't explain myself - could you send me your file please. Thanks John On Thu, 6 Jan 2005 16:41:18 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote: Exactly. I know this is not preferred, but it is the only way I can get around the problem currently (strange state message). I am sure the call progress and busy detect are both no in my conf. I am looking for a "More Correct" answer to this as well. On Jan 6, 2005, at 4:34 PM, John Middleton wrote: What do you mean, as your first priority, you mean exten => s,1,Answer? On Thu, 6 Jan 2005 16:21:35 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote: Yes, I am getting the exact same messages on my console if I do not answer the call as my first priority in the dialplan. On Jan 6, 2005, at 4:11 PM, John Middleton wrote: They are set to No HMMM I get message Ring/off-hook in strange state 6 on channel1 in the log Anyone seen this before On Thu, 6 Jan 2005 23:07:39 +, Phil Quinney <[EMAIL PROTECTED]> wrote: They should be set to no - sorry, I should have been clearer. Busydetect and callprogress cause my X100P's to not answer calls if I have them enabled. Phil. On 6 Jan 2005, at 22:56, John Middleton wrote: what should those two settings say? should i set them to yes, or take the lines out? OK: Do you have these in zapata.conf? busydetect=no callprogress=no They won't work in the UK. ___ 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 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 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 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
Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
Here you go, I was on my home from work: [channels] language=en context=from-pots signalling=fxs_ks usecallerid=yes callerid=asreceived cidsignalling=bell hidecallerid=no busydetect=no callprogress=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 channel => 1 On Jan 6, 2005, at 5:20 PM, John Middleton wrote: and yr zapata.conf please On Thu, 6 Jan 2005 17:01:36 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote: Here is the pertinent section of my extensions.conf: [from-pots] exten => s,1,Answer exten => s,2,Dial(sip/2001&sip/2002&sip/2003,20,t) exten => s,3,Voicemail(u2000) exten => s,103,Voicemail(b2000) exten => s,104,Hangup exten => a,1,VoicemailMain exten => a,2,Hangup exten => t,1,Hangup exten => h,1,Hangup On Jan 6, 2005, at 4:53 PM, John Middleton wrote: Sorry, maybe I didn't explain myself - could you send me your file please. Thanks John On Thu, 6 Jan 2005 16:41:18 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote: Exactly. I know this is not preferred, but it is the only way I can get around the problem currently (strange state message). I am sure the call progress and busy detect are both no in my conf. I am looking for a "More Correct" answer to this as well. On Jan 6, 2005, at 4:34 PM, John Middleton wrote: What do you mean, as your first priority, you mean exten => s,1,Answer? On Thu, 6 Jan 2005 16:21:35 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote: Yes, I am getting the exact same messages on my console if I do not answer the call as my first priority in the dialplan. On Jan 6, 2005, at 4:11 PM, John Middleton wrote: They are set to No HMMM I get message Ring/off-hook in strange state 6 on channel1 in the log Anyone seen this before On Thu, 6 Jan 2005 23:07:39 +, Phil Quinney <[EMAIL PROTECTED]> wrote: They should be set to no - sorry, I should have been clearer. Busydetect and callprogress cause my X100P's to not answer calls if I have them enabled. Phil. On 6 Jan 2005, at 22:56, John Middleton wrote: what should those two settings say? should i set them to yes, or take the lines out? OK: Do you have these in zapata.conf? busydetect=no callprogress=no They won't work in the UK. ___ 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 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 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 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 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
Re: [Asterisk-Users] Broadvoice Status Check 11:18pm PST
Same problem here with dca & chi. 2nd time this week. Ernie On Jan 7, 2005, at 12:18 AM, Mark Halverson wrote: Connecting to proxy.dca.broadvoice.com Calls all of a sudden cannot be completed. Cannot ping proxy.dca.broadvoice.com Changed to proxy.chi.broadvoice.com pingable, but now get the following: Got SIP response 500 "Internal Server Error" back from 147.135.12.128internal error Called Tech Support... get dead air then carrier busy signal. Anyone else having problems? -Mark ___ 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 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] Is this a firefly problem? (*78/*79 doesn't work)
Hello List, On my cvs-head (29-Dec) asterisk, my sip phones can use *78 for DND and *79 to turn it off. With my firefly (iax) client, I am getting the following errors if I try these feature codes: Jan 10 13:26:18 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist Jan 10 13:26:23 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist I do not have any other IAX clients to test this with. Does anyone know if this is just an issue with firefly? Thanks, Ernie ___ 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
Re: [Asterisk-Users] Is this a firefly problem? (*78/*79 doesn't work)
Thanks Paul. I understand and can agree with that. Does that also hold true for a remapped (*8 to *88), PickupGroup? I ask, because this isn't working either from firefly, which is what started the head scratching. *88 is not listed in the regional settings on my sipura 2000's, which is why I chose to use *88, but yet it works via sip. Ernie On Jan 10, 2005, at 1:37 PM, Paul Rodan wrote: I think *78 and *79 are built in features of your IP Phone or Analog converter, not specifically an Asterisk feature, you'd have to rig it in your dial plan. The FireFly software has alternate ways to set do not disturb and call waiting. When you do the *78 or whatever, the DND or Forwarding is done at the phone level, it never gets sent to Asterisk. When you dial *78 with FireFly, since it doesn't have this feature, it sends it to Asterisk, which knows nothing about it. I could be wrong though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernie Ankele Sent: Monday, January 10, 2005 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Is this a firefly problem? (*78/*79 doesn't work) Hello List, On my cvs-head (29-Dec) asterisk, my sip phones can use *78 for DND and *79 to turn it off. With my firefly (iax) client, I am getting the following errors if I try these feature codes: Jan 10 13:26:18 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist Jan 10 13:26:23 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist I do not have any other IAX clients to test this with. Does anyone know if this is just an issue with firefly? Thanks, Ernie ___ 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 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 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
Re: [Asterisk-Users] Is this a firefly problem? (*78/*79 doesn't work)
no *XX in my extensions.conf. In features.conf: [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 1355 ;Number of seconds a call can be parked for ; (default is 45 seconds) pickupexten = *88 ; Configure the pickup extension. Default is *8 I changed pickupexten to *88. (Because the Sipura dialplan already has *XX in it) *78 and *79 are in the regional settings on the sipura, but *88 is not. It appears that Asterisk is handling the *88 correctly for sip phones, but not IAX. Ernie On Jan 10, 2005, at 2:10 PM, Paul Rodan wrote: Is there a *88 in your extensions.conf? Or any * codes for that matter? If it's not listed in the Sipura, it should send it to Asterisk, and if it works, then it has to be in there somewhere. Otherwise it's an "unlisted" or unconfigurable Sipura feature. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernie Ankele Sent: Monday, January 10, 2005 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is this a firefly problem? (*78/*79 doesn't work) Thanks Paul. I understand and can agree with that. Does that also hold true for a remapped (*8 to *88), PickupGroup? I ask, because this isn't working either from firefly, which is what started the head scratching. *88 is not listed in the regional settings on my sipura 2000's, which is why I chose to use *88, but yet it works via sip. Ernie On Jan 10, 2005, at 1:37 PM, Paul Rodan wrote: I think *78 and *79 are built in features of your IP Phone or Analog converter, not specifically an Asterisk feature, you'd have to rig it in your dial plan. The FireFly software has alternate ways to set do not disturb and call waiting. When you do the *78 or whatever, the DND or Forwarding is done at the phone level, it never gets sent to Asterisk. When you dial *78 with FireFly, since it doesn't have this feature, it sends it to Asterisk, which knows nothing about it. I could be wrong though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernie Ankele Sent: Monday, January 10, 2005 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Is this a firefly problem? (*78/*79 doesn't work) Hello List, On my cvs-head (29-Dec) asterisk, my sip phones can use *78 for DND and *79 to turn it off. With my firefly (iax) client, I am getting the following errors if I try these feature codes: Jan 10 13:26:18 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist Jan 10 13:26:23 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist I do not have any other IAX clients to test this with. Does anyone know if this is just an issue with firefly? Thanks, Ernie ___ 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 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 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 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 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] Sip to IAX ok, ZAP to IAX FAILS
Hello, Could someone give me clues where to figure out this problem? If I call from a Sip client to an Firefly client running IAX, the call connects fine, no problems. I can connect to asterisk using any codec (excepting g.729) on firefly to voicemail and music-on-hold, other sip extensions and everything works fine. If I try to connect to the same client via a ZAP channel (X100P clone), via Dial(IAX2/) I get an error : Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in IAX.conf and all codecs are enabled on Firefly. I have tried everything I can think of- only enable gsm, only gsm+G.711, all codecs on firefly. Same results. Anyone else with this issue? Thanks, Ernie ___ 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
Re: [Asterisk-Users] Sip to IAX ok, ZAP to IAX FAILS
On a sip to iax : CODEC_PREFS : (gsm|ulaw|alaw|ilbc) and Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 0ms SCall: 19170 DCall: 1 [xx.xxx.xxx.xxx:20406] FORMAT : 4 -- Call accepted by xx.xxx.xxx.xxx (format ulaw) -- Format for call is ulaw On ZAP to IAX: CODEC_PREFS : (gsm|ulaw|alaw|ilbc) and Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 0ms SCall: 15725 DCall: 3 [xx.xxx.xxx.xxx:20406] CAUSE : No compatible Codecs Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs Thanks, Ernie On Jan 10, 2005, at 6:34 PM, Adam Hart wrote: use ethereal or iax2 debug to see what capabilities are been set in your NEW message Ernie Ankele wrote: Hello, Could someone give me clues where to figure out this problem? If I call from a Sip client to an Firefly client running IAX, the call connects fine, no problems. I can connect to asterisk using any codec (excepting g.729) on firefly to voicemail and music-on-hold, other sip extensions and everything works fine. If I try to connect to the same client via a ZAP channel (X100P clone), via Dial(IAX2/) I get an error : Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in IAX.conf and all codecs are enabled on Firefly. I have tried everything I can think of- only enable gsm, only gsm+G.711, all codecs on firefly. Same results. Anyone else with this issue? Thanks, Ernie ___ 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 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 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
Re: [Asterisk-Users] Sip to IAX ok, ZAP to IAX FAILS
Adam, I think I got it worked out... I changed disallow=723.1 to disallow=all and then accepted back in ulaw,alaw,gsm and ilbc and it started accepting the calls. I do not know why, but its working now. FWIW, here is the full frame as it was before: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 1 DCall: 0 [xx.xxx.xxx.xxx:20406] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (gsm|ulaw|alaw|ilbc) CALLING NUMBER : 3035520218 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 LANGUAGE: en FORMAT : 64 CAPABILITY : 1048575 ADSICPE : 2 DATE TIME : 170564634 thanks, Ernie On Jan 10, 2005, at 7:17 PM, Adam Hart wrote: Can you paste the full NEW frame please. Could be Preference vs capability thanks, Adam Ernie Ankele wrote: On a sip to iax : CODEC_PREFS : (gsm|ulaw|alaw|ilbc) and Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 0ms SCall: 19170 DCall: 1 [xx.xxx.xxx.xxx:20406] FORMAT : 4 -- Call accepted by xx.xxx.xxx.xxx (format ulaw) -- Format for call is ulaw On ZAP to IAX: CODEC_PREFS : (gsm|ulaw|alaw|ilbc) and Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 0ms SCall: 15725 DCall: 3 [xx.xxx.xxx.xxx:20406] CAUSE : No compatible Codecs Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs Thanks, Ernie On Jan 10, 2005, at 6:34 PM, Adam Hart wrote: use ethereal or iax2 debug to see what capabilities are been set in your NEW message Ernie Ankele wrote: Hello, Could someone give me clues where to figure out this problem? If I call from a Sip client to an Firefly client running IAX, the call connects fine, no problems. I can connect to asterisk using any codec (excepting g.729) on firefly to voicemail and music-on-hold, other sip extensions and everything works fine. If I try to connect to the same client via a ZAP channel (X100P clone), via Dial(IAX2/) I get an error : Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in IAX.conf and all codecs are enabled on Firefly. I have tried everything I can think of- only enable gsm, only gsm+G.711, all codecs on firefly. Same results. Anyone else with this issue? Thanks, Ernie ___ 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 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 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 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 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] sip registrations
Hello, I am running 1-11 CVS-HEAD. Per the sip.conf.sample, I should be able to use a register line like: register => username:[EMAIL PROTECTED]/ext if I have [sip_proxy] defined below it. I can't get it to work at all. I keep getting Jan 11 23:47:49 WARNING[1388]: chan_sip.c:1348 create_addr: No such host: broadvoice_in This is my sip.conf sections: register => [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/ xx ... [broadvoice_in] type=user host=sip.broadvoice.com context=from-broadvoice canreinvite=no dtmfmode=inband I am trying this because I would love to get my sip-inbound from different providers in their own contexts. Anybody see what I am doing wrong here? Thanks, Ernie ___ 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
Re: [Asterisk-Users] iconecthere and *
Jer, as I understand it, when you specify /EXTENSION in the register statement, you need to define the EXTENSION in extensions.conf- You can't use "s" for extension matching. If you omit the /EXTENSION in your register statement, then incoming calls will match on "s". I am using iconnecthere, and that is how it seems to be working for me. AS to why you don't see anything in debug sip? I don't know. Ernie On Jan 14, 2005, at 5:46 AM, Jer wrote: Hi all I am trying to figuure out how to get iconnecthere incoming calls to work outbound works fine but incoming goes nowhere but to my iconnecthere vocemail if I do a sip show registry it shows up as regg'ed nnn=is my iconnect here number xxx is my secret Thank you Jeremy [general] qualify=no register=NNN:[EMAIL PROTECTED]/N context=default bind = 0.0.0.0 port=5060 bindaddr=192.168.215.5 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=no videosupport=no disallow=all; First disallow all codecs allow=ulaw relaxdtmf=yes nat=yes ; NAT settings externip = 24.172.122.XXX localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [iconnecthere] type=friend secret= username= I epxect this to be answered by the s ext in the dfault content which is the demo setup a this point or am I missing something all that happens if * never even sees the call (or more to the point i dont see it with sip debug!) THanks Jer ___ 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 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
Re: [Asterisk-Users] Call Pickup
Matt, they work fine on zap and sip. I wish they worked on IAX. Ernie On Jan 24, 2005, at 12:20 PM, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 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
Re: [Asterisk-Users] festival text for weather report
Dean,
Here is a rough starting point for you.
Put this into your /var/lib/asterisk/agi-bin/ directory. I used the
name getforecast. (Change the ZIP code info, line 29) and make it
executable.
<<< - FILE BEGIN ->>>
#!/bin/bash
# I grab the data from asterisk even though I don't use it
while read -e ARG && [ "$ARG" ] ; do
array=(` echo $ARG | sed -e 's/://'`)
export ${array[0]}=${array[1]}
done
checkresults()
{
while read line
do
case ${line:0:4} in
"200 " ) echo $line >&2
return;;
"510 " ) echo $line >&2
return;;
"520 " ) echo $line >&2
return;;
* ) echo $line >&2;;
esac
done
}
# answer the line and give some preliminary feedback
echo "ANSWER "
checkresults
echo "STREAM FILE national-weather-service \"\" "
checkresults
# Grab the forecast info page -- 80003 is MY zipcode, CHANGE TO YOUR
ZIPCODE!
tempstr=` curl -s "http://weather.toolbot.com/?where=80003&RSS"; `
# Cleanup the results, get rid of html tags etc. (Could probably be
condensed)
tempstr=` echo $tempstr | sed 's:
sed 's:<[br /h3]*>::g' | sed 's:nbsp;: :g' | sed
's:&::g' | sed 's:mph:miles per hour:g' `
# Create 'EOF' Mark in tempstr
tempstr=`echo $tempstr "~XOX" `
# Loop through string, echoing to file, convert to wave, speak them,
etc.
until [ "$tempstr" == "XOX" ]
do
lineout=`echo $tempstr | cut -f1 -d"~" `
echo $lineout > /tmp/linetospeak.txt
text2wave -f 8000 -o /tmp/forecastline.wav /tmp/linetospeak.txt
echo "STREAM FILE /tmp/forecastline \"1\""
checkresults
tempstr=` echo $tempstr | cut -f2- -d"~" `
done
echo "STREAM FILE goodbye \"\""
checkresults
rm /tmp/linetospeak.txt
rm /tmp/forecastline.wav
echo "HANGUP "
checkresults
<<<- FILE END ->>>
I have the following in my extensions.conf:
exten => 2996,1,Answer
exten => 2996,2,wait(1)
exten => 2996,3,agi,getforecast
exten => 2996,4,Hangup
I'm not sure how you enter the extension.conf in [EMAIL PROTECTED]
NOTE: I am still LEARNING shell scripting & AGI, so the above may be
kind of hack-ish. Helpful suggestions/advice very welcome!
Ernie Ankele
On Feb 16, 2005, at 9:24 PM, dean collins wrote:
http://www.srh.noaa.gov/fwd/productviewnation.php?
pil=OKXZFPOKX&version=0
can anyone suggest how I could set up [EMAIL PROTECTED] to read out
allowed the following text when I dial extension 850?
815 PM EST WED FEB 16 2005
.OVERNIGHT...MOSTLY CLEAR. LOWS 30 TO 35. NORTHWEST WINDS 15 TO 20
MPH WITH GUSTS UP TO 30 MPH...DIMINISHING TO 10 TO 15 MPH LATE.
.THURSDAY...PARTLY CLOUDY. COOLER WITH HIGHS AROUND 40. NORTHWEST
WINDS AROUND 15 MPH.
.THURSDAY NIGHT...PARTLY CLOUDY. LOWS IN THE MID 20S. WEST WINDS
AROUND 15 MPH.
.FRIDAY...PARTLY CLOUDY AND BRISK. HIGHS IN THE MID 30S. NORTHWEST
WINDS 15 TO 25 MPH.
.FRIDAY NIGHT...PARTLY CLOUDY AND BRISK. LOWS AROUND 17. NORTHWEST
WINDS 15 TO 25 MPH.
There’s $20 via paypal to the first person to help me complete this
(I’ll then post it on the the wiki so anyone can replicate it)
(anyone wanting to add to that bounty email me)
Also if it is not too difficult I’d like it to skip to the next block
each time you press ‘1’ (eg go from overnight to Thursday)
Also it doesn’t need to be this particular web page that it connects
to but something with current weather etc.
Cheers,
Dean
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Re: [Asterisk-Users] festival text for weather report
Dean, No problem, Its actually what I am already using. I just finally
got around to answering the email.
Glad you got it all figured out.
Ernie
On Feb 17, 2005, at 4:07 PM, dean collins wrote:
Hi Ernie,
Man I hope you didn't write all of that for me, I feel really bad now,
someone posted to the list about 15 mins after I posted with the
solution lol- I've already been playing with it for hours working out
what other sites I can get it to read from as well.
Thanks anyway - good practive I guess.
I'm modifying the festival wikki page once I work out how.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernie
Ankele
Sent: Thursday, February 17, 2005 5:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] festival text for weather report
Dean,
Here is a rough starting point for you.
Put this into your /var/lib/asterisk/agi-bin/ directory. I used the
name getforecast. (Change the ZIP code info, line 29) and make it
executable.
<<< - FILE BEGIN ->>>
#!/bin/bash
# I grab the data from asterisk even though I don't use it
while read -e ARG && [ "$ARG" ] ; do
array=(` echo $ARG | sed -e 's/://'`)
export ${array[0]}=${array[1]}
done
checkresults()
{
while read line
do
case ${line:0:4} in
"200 " ) echo $line >&2
return;;
"510 " ) echo $line >&2
return;;
"520 " ) echo $line >&2
return;;
* ) echo $line >&2;;
esac
done
}
# answer the line and give some preliminary feedback
echo "ANSWER "
checkresults
echo "STREAM FILE national-weather-service \"\" "
checkresults
# Grab the forecast info page -- 80003 is MY zipcode, CHANGE TO YOUR
ZIPCODE!
tempstr=` curl -s "http://weather.toolbot.com/?where=80003&RSS"; `
# Cleanup the results, get rid of html tags etc. (Could probably be
condensed)
tempstr=` echo $tempstr | sed 's:
| sed 's:<br />:~:g'|\
sed 's:<[br /h3]*>::g' | sed 's:nbsp;: :g' | sed
's:&::g' | sed 's:mph:miles per hour:g' `
# Create 'EOF' Mark in tempstr
tempstr=`echo $tempstr "~XOX" `
# Loop through string, echoing to file, convert to wave, speak them,
etc.
until [ "$tempstr" == "XOX" ]
do
lineout=`echo $tempstr | cut -f1 -d"~" `
echo $lineout > /tmp/linetospeak.txt
text2wave -f 8000 -o /tmp/forecastline.wav
/tmp/linetospeak.txt
echo "STREAM FILE /tmp/forecastline \"1\""
checkresults
tempstr=` echo $tempstr | cut -f2- -d"~" `
done
echo "STREAM FILE goodbye \"\""
checkresults
rm /tmp/linetospeak.txt
rm /tmp/forecastline.wav
echo "HANGUP "
checkresults
<<<- FILE END ->>>
I have the following in my extensions.conf:
exten => 2996,1,Answer
exten => 2996,2,wait(1)
exten => 2996,3,agi,getforecast
exten => 2996,4,Hangup
I'm not sure how you enter the extension.conf in [EMAIL PROTECTED]
NOTE: I am still LEARNING shell scripting & AGI, so the above may be
kind of hack-ish. Helpful suggestions/advice very welcome!
Ernie Ankele
On Feb 16, 2005, at 9:24 PM, dean collins wrote:
http://www.srh.noaa.gov/fwd/productviewnation.php?
pil=OKXZFPOKX&version=0
can anyone suggest how I could set up [EMAIL PROTECTED] to read out
allowed the following text when I dial extension 850?
815 PM EST WED FEB 16 2005
.OVERNIGHT...MOSTLY CLEAR. LOWS 30 TO 35. NORTHWEST WINDS 15 TO 20
MPH WITH GUSTS UP TO 30 MPH...DIMINISHING TO 10 TO 15 MPH LATE.
.THURSDAY...PARTLY CLOUDY. COOLER WITH HIGHS AROUND 40. NORTHWEST
WINDS AROUND 15 MPH.
.THURSDAY NIGHT...PARTLY CLOUDY. LOWS IN THE MID 20S. WEST WINDS
AROUND 15 MPH.
.FRIDAY...PARTLY CLOUDY AND BRISK. HIGHS IN THE MID 30S. NORTHWEST
WINDS 15 TO 25 MPH.
.FRIDAY NIGHT...PARTLY CLOUDY AND BRISK. LOWS AROUND 17. NORTHWEST
WINDS 15 TO 25 MPH.
There's $20 via paypal to the first person to help me complete this
(I'll then post it on the the wiki so anyone can replicate it)
(anyone wanting to add to that bounty email me)
Also if it is not too difficult I'd like it to skip to the next block
each time you press '1' (eg go from overnight to Thursday)
Also it doesn't need to be this particular web page that it connects
to but something with current weather etc.
Cheers,
Dean
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Re: [Asterisk-Users] weather asterisk@home
Turn on debugging (agi debug) and check to see if festival is exiting
with an error? (Maybe)
Ernie
On Feb 25, 2005, at 4:29 PM, James Taylor wrote:
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
-- Executing Answer("SIP/3000-a844", "") in new stack
-- Executing AGI("SIP/3000-a844", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-a844", "") in new stack
== Spawn extension (from-internal, 850, 3) exited non-zero on
'SIP/3000-a844'
-- Executing Macro("SIP/3000-a844", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-a844", "w") in new stack
-- Executing NoCDR("SIP/3000-a844", "") in new stack
-- Executing Wait("SIP/3000-a844", "5") in new stack
-- Executing Hangup("SIP/3000-a844", "") in new stack
Any ideas?
--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas 75503
903-793-1953
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Re: [Asterisk-Users] weather asterisk@home
My perl is not that great, but from your debug output, the "STREAM
FILE" agi command never executed from the festival-weather-script.pl
script, which should have happened right away.
Does your text2wave work? Is there a tts-.wav file in your
var/lib/asterisk/sounds/tts dir? a tts-???.txt file?
You should be able to use this info to find out where the script is
failing.
Ernie
On Feb 25, 2005, at 5:25 PM, James Taylor wrote:
Still no weather...
AGI Debugging Enabled
-- Executing Answer("SIP/3000-51a3", "") in new stack
-- Executing AGI("SIP/3000-51a3", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
AGI Tx >> agi_request: weather.agi
AGI Tx >> agi_channel: SIP/3000-51a3
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1109375983.318
AGI Tx >> agi_callerid: "James" <3000>
AGI Tx >> agi_dnid: 850
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: from-internal
AGI Tx >> agi_extension: 850
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-51a3", "") in new stack
== Spawn extension (from-internal, 850, 3) exited non-zero on
'SIP/3000-51a3'
-- Executing Macro("SIP/3000-51a3", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-51a3", "w") in new stack
-- Executing NoCDR("SIP/3000-51a3", "") in new stack
-- Executing Wait("SIP/3000-51a3", "5") in new stack
-- Executing Hangup("SIP/3000-51a3", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/3000-51a3' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
On Fri, 25 Feb 2005 16:31:43 -0700, Ernie Ankele <[EMAIL PROTECTED]>
wrote:
Turn on debugging (agi debug) and check to see if festival is exiting
with an error? (Maybe)
Ernie
On Feb 25, 2005, at 4:29 PM, James Taylor wrote:
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
-- Executing Answer("SIP/3000-a844", "") in new stack
-- Executing AGI("SIP/3000-a844", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-a844", "") in new stack
== Spawn extension (from-internal, 850, 3) exited non-zero on
'SIP/3000-a844'
-- Executing Macro("SIP/3000-a844", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-a844", "w") in new stack
-- Executing NoCDR("SIP/3000-a844", "") in new stack
-- Executing Wait("SIP/3000-a844", "5") in new stack
-- Executing Hangup("SIP/3000-a844", "") in new stack
Any ideas?
-- James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas 75503
903-793-1953
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--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas 75503
903-793-1953
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