[asterisk-users] Problem with Asterisk 1.4 and Linksys Spa941/962
Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the phone hangs up the call after its connected to the other side. Sip debug shows me the following scenario: - invite - 407 proxy authorisation - ACK - Invite with auth header - 100 trying - 183 session progress with sdp header (correct c and m header) - 200 OK with same sdp header - ACK - BYE - 200 OK this also happens on music on hold or playback and when trying to bridge 2 channels. this only happens on one location and several (1000) clients with the same phone had no problem. also a snom360 and xlite could dial out without any problem in the same network. After we had downgrade to 1.2.32 everything works fine again on these phones. my question is, had there been a big change in sip.conf or codec handling which cause this problem, cause i used the same sip.conf just adding notifyringing=yes, limitonpeers=yes and allowsubscribe=yes. Here is my sip.conf with one client: [general] context=incoming realm=softpbx bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useclientcode=yes defaultexpirey=3600 vmexten=voicemail disallow=all allow=alaw allow=ulaw allow=gsm ;qualify=no ;canreinvite=no musicclass=default language=de useragent=ipanlage callevents=yes nat=yes rtcachefriends=no rtupdate=no rtautoclear=no ignoreregexpire=yes amaflags=omit canreinvite=no subscribecontext=outcust limitonpeers=yes allowsubscribe=yes notifyringing=yes [xxx] type=friend context=outcust nat=yes qualify=yes secret= username=xxx callerid=bla bla accountcode=xxx disallow=all allow=alaw allow=ulaw allow=gsm host=dynamic best regards steve smith -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with asterisk 1.4 Installation (undefined reference to `ast_copy_string')
Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c - ast_expr2f.o [CC] ast_expr2.c - ast_expr2.o [CC] strcompat.c - strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o - aelparse aelparse.o(.text+0x3029): In function `ael_yylex': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ael.flex:417: undefined reference to `ast_copy_string' ast_expr2f.o(.text+0x1198): In function `ast_expr': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ast_expr2.fl:248: undefined reference to `ast_copy_string' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error... plz could answer this issue. -nsthi ___ --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] Problem with asterisk 1.4 Installation (undefined reference to `ast_copy_string')
Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c - ast_expr2f.o [CC] ast_expr2.c - ast_expr2.o [CC] strcompat.c - strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o - aelparse aelparse.o(.text+0x3029): In function `ael_yylex': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ael.flex:417: undefined reference to `ast_copy_string' ast_expr2f.o(.text+0x1198): In function `ast_expr': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ast_expr2.fl:248: undefined reference to `ast_copy_string' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error... plz could answer this issue. -nsthi ___ --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] problem with asterisk 1.4
Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi ___ --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] problem with asterisk-1.4+sip communicator
Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi ___ --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] problem with asterisk-1.4+sip communicator
On 12/6/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi Hi is the both users are behind NAT ? Ram ___ --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] problem with asterisk-1.4+sip communicator
hi, i'm using same network for asterisk server ,PBX agent and aslo users ...like an office have 20 PCs.. and my present sip.conf nat=no.is i need to set it as yes or any suggestions.. regards, thiru On 12/7/06, ram [EMAIL PROTECTED] wrote: On 12/6/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi Hi is the both users are behind NAT ? Ram ___ --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
