RE: [Asterisk-Users] Asterisk .call files
Hi is the syntax of this call file correct? because wheniit to /var/spool/asterisk/outgoing, the CLI shows"unknown keyword" for all the keywords used (i.e. channel, MaxRetries,...). 1.call Channel:Zap/g2/5148367580 MaxRetries:2 RetryTime:60 WaitTime:30Context:extensions Extension:1234Priority:1 Regards, ___ 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] Asterisk .call files
hi I created a .call file as mentioned in the WiKi but when i place it in /var/spool/asterisk/outgoing, the Asterisk console shows unknown keyword for all the keywords used in the .call file (i.e channel, context, extension,...). Any ideas why? Regards, Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490 ___ 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] Cisco 7940 convert to sip
Hi This is found on cisco.com. http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml hope it helps Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Krasavin Andrey Sent: Friday, March 18, 2005 7:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7940 convert to sip Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar 18 15:12:30 AKrasavin xinetd[10068]: EXIT: tftp pid=10080 duration=0(sec) Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10082 from=192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10083]: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10083]: peer requests SEP000AF4BB7D59.cnf.xml, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10083]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: EXIT: tftp pid=10082 duration=0(sec) OS79XX.TXT and SEP000AF4BB7D59.cnf.xml exists. I'll be very thankful for any your help. -- WBR, Krasavin Andrey ___ 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: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
Hi I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through SIP. Can you please send me the Dial-peer configuration that creates a trunk between the Cisco router and Asterisk. Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Jones Sent: Wednesday, March 16, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Possible SPAM] :[Asterisk-Users] about sip,asterisk and cisco ccme I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a call manager extension, and using the dial-plan to direct the call to *, and I do successfully get the * voicemail. Why do you want to use h323/skinny rather than SIP? -Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Riela Sent: Wednesday, March 16, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I would create a structure like this: external sip server \ external sip server |-| Asterisk |--| Cisco CME |---| ip phones | external sip server / I would use Asterisk as SIP client for some SIP accounts on external servers ... then register those via H323 (if possible; skynny?) on Cisco CME ... Then I would use Asterisk to add the voicemail feature to Cisco CME. I don't know if that's possible, I'm really newbie on Asterisk, I know only Cisco world, and just a little bit. Any advice will be appreciated. Thanks for your support Regards dott. Andrea Riela -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCOLJtMakHrsrHP9wRAmDfAJ9AgcMf1CmdrLBk4HEdlvWKZiht7QCfcgns GbTX2LxGxO3ZR7iMIPqreJA= =eKlT -END PGP SIGNATURE- ___ 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] Asterisk support for SIP REFER message
Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the transfer tag and setting bridge=true (i.e transfer name=transfer1 bridge=true connecttimeout=10s ) but as soon as Asterisk receives the SIP REFER message generated by the VoiceXML application, it sends back a NOTIFY message with a subscription-state:terminated as if it was a blind transfer (bridge=false) which instructs the VoiceXML application to disconnect so it no longer supervises the call to get back the result ( callee unavailable, busy,...) . Usually, when the brige=true is set in the VoiceXML application, the end point that receives the SIP REFER should send a NOTIFY message with subscription-state:active and then it should send back NOTIFY messages to tell the VoiceXML application about the result of the call (i.e callee unavailable, busy,...). Regards, Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490 ___ 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, DNIS, Asterisk
you can extract the ANI and DNIS from the SIP INVITE message when you call your IVR. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jochen Witte Sent: Thursday, March 10, 2005 3:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP, DNIS, Asterisk Hello, I am currently evaluating the possibilities to change our telephony infrastructure for an IVR system to VoIP. I would like to try out Asterisk like this: PSTN Asterisk - IVR ISDN SIP/RTP My main question is: Is it possible to forward the ANIS to my IVR via SIP? If it is: how to I achieve this? Regards Jochen -- Jochen Witte [EMAIL PROTECTED] ___ 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, DNIS, Asterisk
yep, you can find the ANI and DNIS in the To: and From: fields of the SIP INVITE message. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jochen Witte Sent: Thursday, March 10, 2005 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP, DNIS, Asterisk OK - just to be sure. When using Atserisk as a VoIP gateway, the incoming ISDn ANI (e.g. 123456) will be rewritten to something like sip:[EMAIL PROTECTED] ? Am Donnerstag, den 10.03.2005, 15:48 -0500 schrieb Gilbert Abboud: you can extract the ANI and DNIS from the SIP INVITE message when you call your IVR. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jochen Witte Sent: Thursday, March 10, 2005 3:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP, DNIS, Asterisk Hello, I am currently evaluating the possibilities to change our telephony infrastructure for an IVR system to VoIP. I would like to try out Asterisk like this: PSTN Asterisk - IVR ISDN SIP/RTP My main question is: Is it possible to forward the ANIS to my IVR via SIP? If it is: how to I achieve this? Regards Jochen -- Jochen Witte [EMAIL PROTECTED] ___ 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] Call Progress Analysis
Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and executes the command there as if nobody answered the call. I need this because I want to have a follow-me application that dials different phone numbers or extensions based on the call status. Thank you in advance for your help. Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490 ___ 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] Support for SIP REFER message
Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the Transfer element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the transfer element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with 'subscription-state: terminated to the VoiceXML application as if it was a blind transfer (bridge=false). Usually, when 'bridge=true' is set in the transfer element, the end point that receives the SIP REFER (i.e Asterisk) should send a SIP NOTIFY to the VoiceXML application with 'subscription-state: active' to let it supervise the call and then the endpoint should send Follow-On NOTIFY messages to let the VoiceXML application know the status of the call (no answer,busy,...). Thank you in advance for your help. Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490 ___ 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
