RE: [Asterisk-Users] Avaya 4610sw SIP setup problem
Hi, You can call me by my first name (Silviu) :)) I have made the changes to the settings file, I have removed the LDAP-related settings - nothing changes... The file is still taken into account, as other changes affect the phone, but the SIP fields stay desperately blank... I don't think I'll wait for the next firmware release, I'm currently evaluating several Siemens optiPoint phones (SIP) which look good so far. I have to get things moving, the customer won't wait forever for the Avaya phones to work.c However I'm a bit disappointed to leave things as they are, I have a feeling of ... failure? I guess I'll still try some thing or another in my (inexistent) spare time. Thanks for your help, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 04 July 2006 03:57To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Herchi,I want you to re-read my last e-mail very carefully. Your response does not mention at all my guess that the three SP_DIRSRVR variables may be giving you trouble. I'm still interested in knowing what happens if you remove them from your settings file. Also, I have heard a rumour that there will be a new firmware release on July 10th. Actually, I just clicked the feedback button on their web page for the firmware download and asked. They responded on the first business day (unusual for Avaya), indicating 7/10 is the approximate release date. So there you have my source Let me know On 7/3/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi, I had edited out all lines starting with a #, which is ot right, as the marker for comments is##... See below for the entire file. I just tried the configuration throughDHCP, by setting the 176 option to point to the right TFTP server and also to the right SIP proxy. The Avaya boot test application is not complaining, but the phones ... do I need to say it? *sigh* SET DOMAIN "company.com" SET DNSSRVR "204.140.111.43"SET PHNCC "352"SET PHNDPLENGTH "4"SET PHNIC "00"SET PHNOL "0"SET SYSLANG "English"SET APPSTAT "1"SET RESTORESTAT "1"SET AGCHAND "0"SET AGCHEAD "0"SET AGCSPKR "0"SET SNTPSRVR "204.140.111.200"SET DSTOFFSET "1"SET DSTSTART "1SunApr2L"SET DSTSTOP "LSunOct2L"SET GMTOFFSET "-5:00"SET DATESEPARATOR "/"SET DATETIMEFORMAT "3" SET SIPDOMAIN "slt05.company.agn" SET SIPPROXYSRVR "204.140.111.219"SET SIPPORT "5070" SET SIPREGISTRAR "204.140.111.219" SET DIALPLAN "[234]xxx|55"SET DIALWAIT "3"SET MUSICSRVR ""SET MWISRVR ""SET PHNNUMOFSA "3"SET REGISTERWAIT "120" SET SP_DIRSRVR "10.1.1.1"SET SP_DIRSRVRPORT "389"SET SP_DIRTOPDN "ou=People,o=avaya.com"IF $MODEL4 SEQ 4602 goto SETTINGS4602IF $MODEL4 SEQ 4610 goto SETTINGS4610IF $MODEL4 SEQ 4620 goto SETTINGS4620IF $MODEL4 SEQ 4621 goto SETTINGS4621IF $MODEL4 SEQ 4622 goto SETTINGS4622IF $MODEL4 SEQ 4625 goto SETTINGS4625IF $MODEL4 SEQ 4630 goto SETTINGS4630goto END # SETTINGS4602goto END# SETTINGS4610 SET WMLHOME " http://support.avaya.com/elmodocs2/avayaip/4620/home.wml" SET WMLPROXY "204.140.111.246" SET WMLPORT "3128"goto END # SETTINGS4620goto END# SETTINGS4621goto END# SETTINGS4622goto END# SETTINGS4625goto END# SETTINGS4630 SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htmSET PHNEMERGNUM 112goto END # END From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom Lynn Sent: 01 July 2006 18:18 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are removed? SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com I can't find these three entries anywhere in my 46xx settings file.I also cannot find them in the lan admin guide from the manufacturer.They seem to be somewhat like the ldap options for the 4630 phone, but those didn't have a leading SP_ prefix on the variable name. Why don't you comment them out and see what happens?Tom Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4
RE: [Asterisk-Users] Avaya 4610sw SIP setup problem
Hi, I had edited out all lines starting with a #, which is ot right, as the marker for comments is##... See below for the entire file. I just tried the configuration throughDHCP, by setting the 176 option to point to the right TFTP server and also to the right SIP proxy. The Avaya boot test application is not complaining, but the phones ... do I need to say it? *sigh* SET DOMAIN "company.com"SET DNSSRVR "204.140.111.43"SET PHNCC "352"SET PHNDPLENGTH "4"SET PHNIC "00"SET PHNOL "0"SET SYSLANG "English"SET APPSTAT "1"SET RESTORESTAT "1"SET AGCHAND "0"SET AGCHEAD "0"SET AGCSPKR "0"SET SNTPSRVR "204.140.111.200"SET DSTOFFSET "1"SET DSTSTART "1SunApr2L"SET DSTSTOP "LSunOct2L"SET GMTOFFSET "-5:00"SET DATESEPARATOR "/"SET DATETIMEFORMAT "3"SET SIPDOMAIN "slt05.company.agn"SET SIPPROXYSRVR "204.140.111.219"SET SIPPORT "5070"SET SIPREGISTRAR "204.140.111.219"SET DIALPLAN "[234]xxx|55"SET DIALWAIT "3"SET MUSICSRVR ""SET MWISRVR ""SET PHNNUMOFSA "3"SET REGISTERWAIT "120"SET SP_DIRSRVR "10.1.1.1"SET SP_DIRSRVRPORT "389"SET SP_DIRTOPDN "ou=People,o=avaya.com"IF $MODEL4 SEQ 4602 goto SETTINGS4602IF $MODEL4 SEQ 4610 goto SETTINGS4610IF $MODEL4 SEQ 4620 goto SETTINGS4620IF $MODEL4 SEQ 4621 goto SETTINGS4621IF $MODEL4 SEQ 4622 goto SETTINGS4622IF $MODEL4 SEQ 4625 goto SETTINGS4625IF $MODEL4 SEQ 4630 goto SETTINGS4630goto END# SETTINGS4602goto END# SETTINGS4610SET WMLHOME "http://support.avaya.com/elmodocs2/avayaip/4620/home.wml"SET WMLPROXY "204.140.111.246"SET WMLPORT "3128"goto END# SETTINGS4620goto END# SETTINGS4621goto END# SETTINGS4622goto END# SETTINGS4625goto END# SETTINGS4630SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htmSET PHNEMERGNUM 112goto END# END From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 01 July 2006 18:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are removed? SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com I can't find these three entries anywhere in my 46xx settings file.I also cannot find them in the lan admin guide from the manufacturer.They seem to be somewhat like the ldap options for the 4630 phone, but those didn't have a leading SP_ prefix on the variable name. Why don't you comment them out and see what happens?Tom Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
RE: [Asterisk-Users] Avaya 4610sw SIP setup problem
Hi I tried that too, but the only useful thing I can change (besides the IP settings of the phone itself) is the "CallSv" parameter; I set it to the IP of the SIP registrar/proxy but it still doesn't work... Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HenkSent: 29 June 2006 21:16To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Avaya 4610sw SIP setup problem Did you try to manually to change the parameters of the phone? When you power the phone up then are you able to enter manually the parameter when you hit *. I am using a 4610 with Release 2.2 but I am not using the capability to upload the settings from the server. Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herchi SilviuSent: donderdag 29 juni 2006 15:55To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Avaya 4610sw SIP setup problem I just tried serving the files off Apache, port 80, no change... Most parameters are taken into account by the phone, except for SIP proxy and SIP registrar... Coud someone post an excerpt from their 46xxsettings.txt where I could see the format they use? Thank you in advance, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 29 June 2006 00:33To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN " sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219 " SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END __
RE: [Asterisk-Users] Avaya 4610sw SIP setup problem
I just tried serving the files off Apache, port 80, no change... Most parameters are taken into account by the phone, except for SIP proxy and SIP registrar... Coud someone post an excerpt from their 46xxsettings.txt where I could see the format they use? Thank you in advance, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 29 June 2006 00:33To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN " sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219 " SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
RE: [Asterisk-Users] Avaya 4610sw SIP setup problem
Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
RE: [Asterisk-Users] HDLC Bad FCS (8)
Hi, Take a look here: http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.htmlit might help. Otherwise you can also try different settings for the "span" line in zaptel.conf Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué ContiSent: 28 June 2006 12:33To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] HDLC Bad FCS (8) Hi All. Somebody of you already passed below for this error? Jun 28 02:25:03 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 02:52:08 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 04:11:10 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 04:29:29 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span1 I am not detectingfails in link, have one asterisk-1.2.9.1 linked with a central office Siemens HiPath 4000 and I believe it is functioning, although the times the call to be completed without Ring, nor audio. I hug to all Best Regards Josué ___ --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] Avaya 4610sw SIP setup problem
Title: Avaya 4610sw SIP setup problem Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying Registering for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070(this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya.com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___ --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] Re: siemens pbx and asterisk
Title: Re: siemens pbx and asterisk Hi Lito, We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers. On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an H.323 channel (asterisk-oh323 works best for us). On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk. I hope this helps, Silviu --- Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito ___ --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] Re: Asterisk x Siemens HiPath 4000
Title: Re: Asterisk x Siemens HiPath 4000 Hi, Could you post your /etc/zaptel.conf and zapata.conf? Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)? Silviu Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___ --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
RE: [Asterisk-Users] Re: siemens pbx and asterisk
Hi, As I wrote, the HiPath needs to be upgraded to version 3 (don't ask me any details, I'm not a Siemens expert) in order to have the CallerID name passed over the H.323 link. Earlier versions (my case) ony sends and accepts the CallerId number. I have set up a workaround for calls coming to Asterisk: an AGI script sets the CallerID name according to their CallerID number by looking it up in a database. This is done in real time for every incoming call. Obviously it doesn't work for calls going from Asterisk to the HiPath. Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Hamann Sent: 27 June 2006 14:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: siemens pbx and asterisk Hi Silviu, did you manage to get the callername to work? I have a comparable setup with a hipath System but I can´t get the callername to be displayed over the trunk. The callernumber works but not the name... Any suggestion? Thanks Michael We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers. On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an H.323 channel (asterisk-oh323 works best for us). On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk. I hope this helps, Silviu --- Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito ___ --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 ___ --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
RE: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000
Hello, The main differences I can see: - in zaptel.conf you have span=1,0,0,ccs,hdb3, which means you ask Asterisk to serve as a timer for the PBX - on my setup the PBX is the master clock and Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4) - in zapata.conf I have switchtype=EuroISDN. Generally speaking, try using other switchtypes. Regards, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué ContiSent: 27 June 2006 14:41To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000 Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf #zapte.conf span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us #zapata.conf [trunkgroups] [channels]language=pt_BRcontext=defaultswitchtype=qsigpridialplan=privateprilocaldialplan=privatefacilityenable = yessignalling=pri_cpe;rxwink=300usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes rxgain=0.0txgain=0.0group=1callgroup=1immediate=nocallerid=asreceivedmusiconhold=defaultgroup=1channel=1-15channel=17-31 Best Regards Josué 2006/6/27, Herchi Silviu [EMAIL PROTECTED]: Hi, Could you post your /etc/zaptel.conf and zapata.conf? Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)? Silviu Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Meetme and authentication
Title: Meetme and authentication Hi all, I have thoroughly read the available documentation and I can't seem to find a workaround for my setup I'm trying to create a phone conference line that users would call using a unique phone number (no matter if they are moderators or just plain users). I use Asterisk 1.2.6 The available conferences are defined as follows: conf = 1000,user pin1, moderator pin1 conf = 1001,user pin2, moderator pin2 conf = 1002,user pin3, moderator pin3 conf = 1009, user pin9, moderator pin9 The users are prompted whether they are a moderator or a user. When they choose, they are redirected to the conference they request: - using options aAPsX for moderators (moderator + marked + ask PIN + allow menu using *) - using options Psw for users (ask PIN + allow menu + wait for a marked user) My problem is that if a user chooses the moderator option, he can authenticate using any of the two PINs, and he can become an moderator for the conference by knowing only the user PIN I think using two different phone numbers (one for users and one for moderators) is neither practical nor safe. Is there a way to authenticate users against only one of the password? For instance, math the password provided against only the moderator PIN, or only the user PIN. Thank you for your help, Silviu PS. Here is the dialplan : [ConfStart] exten = s,1,Answer exten = s,2,Set(TIMEOUT(response)=5) exten = s,3,Set(LANGUAGE()=conf) exten = s,4,Wait(1) exten = s,5,Background(welcome) ; welcome, press * if you are a user of hold the line if you are a moderator exten = *,1,MeetMe(|iMPsw|) ; for regular users exten = t,1,MeetMe(|aAiMPsX|) ; for moderators exten = i,1,GoTo(ConfStart,s,1) ___ --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
RE: [Asterisk-Users] correct version of asterisk for oh323
Hello, Can you post your oh323.conf? Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] correct version of asterisk for oh323 Hi Herci, I have tried this. pwlib, openh323 and Asterisk-OH323 0.7.3 compiled with no problems. But when you start asterisk, Apr 18 17:47:39 ERROR[11385]: chan_oh323.c:5353 load_module: H.323 listener creation failed. Apr 18 17:47:39 WARNING[11385]: loader.c:414 __load_resource: chan_oh323.so: load_module failed, returning -1 == Cleaning up OpenH323 channel driver. Apr 18 17:47:39 WARNING[11385]: loader.c:554 load_modules: Loading module chan_oh323.so failed! I am using FC3 with 2.6.5-1.358 kernel. Any suggestions? yusuf Herchi Silviu wrote: Hello, I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch versions of OpenH323 and Pwlib (available on http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php). Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] correct version of asterisk for oh323 Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. ___ --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 ___ --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
RE: [Asterisk-Users] IVR: playing multiple streams simultaneously?
The problem with this solution is that the IVR uses phrases generated on the fly using pre-recorded words and digits. So I can not pre-mix the music, it has to be sent along. Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: mercredi 19 avril 2006 09:53To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] IVR: playing multiple streams simultaneously? hi i don't know if we can do that . but i guess we can use audacity .. to mix both the files and get what you want.-Giridhar Bandi On 4/18/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help, Silviu ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
RE: [Asterisk-Users] Remember the incoming context?
Hi
Have you tried using something like
Set(ORIGINAL_CHANNEL=from-sip)
in the original channel?
You can then use Dial(Local/number/${ORIGINAL_CHANNEL}).
Regards,
Silviu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edwin
Groothuis
Sent: mercredi 19 avril 2006 00:57
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Remember the incoming context?
Greetings,
Somewhere on my asterisk system, a calls come in in a certain context,
for example, from-sip or from-pstn.
Then the calls gets routed through the dialplan, and a macro gets
called, and another one and then the call needs to be redirected to
another number in the same initial context. And you can use
Dial(Local/number/initialcontext) for that.
Oops, this initial context is lost somewhere on the line.
Unless I'm very mistaken, there is no way to find out what the original
context was, is there?
Edwin
--
Edwin Groothuis |Personal website:
http://www.mavetju.org
[EMAIL PROTECTED]| Weblog:
http://weblog.barnet.com.au/edwin/
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RE: [Asterisk-Users] Receiving Faxes...
Hello,
I think you should handle the fax in the h (for Hangup) extension (which is,
after your fax was received), instead of using the priorities following the fax
reception (as in your example). Have a look at the different examples in the
wiki, like http://www.voip-info.org/wiki-Asterisk+fax:
[fax]
exten = 666,1,Macro(faxreceive)
exten = h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} ${CALLERIDNUM})
Regards,
Silviu
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Gröger
Sent: mardi 18 avril 2006 21:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Receiving Faxes...
Hi,
I am experimenting with receiving faxes in asterisk:
exten = in_fax,1,Macro(faxreceive)
exten = in_fax,2,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf
- ${FAXFILE}.pdf)
exten = in_fax,3,system(cp ${FAXFILE}.pdf
/var/www/faxes/${CALLERID(number)}.pdf)
exten = in_fax,4,system(mime-construct --to [EMAIL PROTECTED] --subject Fax
from ${CALLERID(number)} ${CALLERID(name)} --attachment
${CALLERID(number)}.pdf --type application/pdf --file ${FAXFILE}.pdf)
exten = in_fax,5,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten = in_fax,6,Hangup
That is an extension Freepbx made, with some extensions from me, because
FreePBX doesn't work well with mISDN... Wel, it receives faxes and it
saves them as an tif, it also converts them to a pdf file, but the other
commands aren't executed... why?
thanks for help
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[Asterisk-Users] IVR: playing multiple streams simultaneously?
Title: IVR: playing multiple streams simultaneously? Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help, Silviu ___ --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
RE: [Asterisk-Users] correct version of asterisk for oh323
Hello, I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch versions of OpenH323 and Pwlib (available on http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php). Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] correct version of asterisk for oh323 Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf ___ --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 ___ --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] CDR manipulation in macros
Hi all,
I'm trying to change the CDR userfield in a macro which is executed upon
call pickup (option M in Dial command). The goal is to log the answer
time (in the default CDR it is not correct as the call is picked up to
play music on hold to the caller before Dialing the called extension). I
use Asterisk 1.0.9, with asterisk-oh323 0.6.5.
Here is my dialplan:
...
exten = s,8,Dial(OH323/[EMAIL PROTECTED]:1720,20,mM(CdrAnswerDate)) ;
execute macro-CdrAnswerDate when the called extension 1234 is picked up
exten = s,9,AppendCDRUserField(no_answer ) ; if no answer after 20
sec.
...
The macro-CdrAnswerDate is defined as follows:
[macro-CdrAnswerDate]
exten = s,1,AGI(getCurrentTimeDate.sh) ; shell script that sets the
variable ANSWER_DATE to the pickup date
exten = s,2,AppendCDRUserField(answered ${ANSWER_DATE})
Here is what I get in the console:
-- Started music on hold, class 'default', on
OH323/HiPath4000,@10.253.3.27-393a
H.323 call 'ip$localhost/18575', exception CALL_ALERTED.
-- OH323/[EMAIL PROTECTED] is ringing
H.323 call 'ip$localhost/18575', exception CALL_ESTABLISHED.
-- OH323/[EMAIL PROTECTED] answered
OH323/HiPath4000,@10.253.3.27-393a
-- Executing AGI(OH323/[EMAIL PROTECTED],
getCurrentTimeDate.sh) in new stack
-- Launched AGI Script /usr/local/asterisk/agi/getCurrentTimeDate.sh
getCurrentTimeDate.sh: Call answered 2005-12-08 11:04:51
-- AGI Script getCurrentTimeDate.sh completed, returning 0
-- Executing AppendCDRUserField(OH323/[EMAIL PROTECTED],
answered 2005-12-08 11:04:51) in new stack
-- Stopped music on hold on OH323/HiPath4000,@10.253.3.27-393a
-- H.323 call 'ip$localhost/18575' cleared, reason 4 (Cleared by
remote user), established (2 sec)
-- Hungup 'OH323/[EMAIL PROTECTED]'
Which seems to indicate that it is OK.
However the recorded CDR Userfield (I use MySQL for that) is not
updated: it contains only the values I had Append-ed before...
Is there a problem with changing CDRs in macros? My previous tests
showed that using ForkCDR or ResetCDR in macros doesn't work either.
Theank you for your help.
Regards,
Silviu
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