RE: [asterisk-users] Interconnecting Cisco 1760 routers with Asterisk
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_
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joesph
Enviado el: Martes, 03 de Abril de 2007 02:53 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] Interconnecting Cisco 1760 routers with Asterisk
Good day everyone.
I have Cisco 1760 routers that do site to site voip. Each router has 2 fxs
ports that connect to the local pbx and use sip to connect to other routers
over the WAN. I am thinking of putting in an asterisk box at the hub site
for interconnectivity with our global office voip provider. This provider
runs asterisk.
Question is - can Cisco 1760 routers make/receive calls to/fro asterisk? if
yes, any sample configuration please?
Thanks and regards
Joesph
Abuja, Nigeria
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RE: [asterisk-users] Re: Marks SNMP HowTo
How can i see if snmp is running ok on mi * box ? Thanks in advance -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Forrest Beck Enviado el: Domingo, 25 de Febrero de 2007 06:14 p.m. Para: Asterisk Users List Asunto: [asterisk-users] Re: Marks SNMP HowTo OK. problem solved. It was something dumb on my part. /var/agentx didn't have enough permissions to let asterisk access the socket. On 2/25/07, Forrest Beck [EMAIL PROTECTED] wrote: I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is doing. Does it bind to a particular port that maybe my asterisk user does not have permission to access??? Here is my snmpd.conf file: master agentx agentXPerms 0660 0550 asterisk asterisk com2sec local localhost da_public com2sec mynetwork 10.11.0.0/16 da_public com2sec dmz 172.17.0.0/16 da_public group MyROGroup any local group MyROGroup any mynetwork group MyROGroup any dmz view all included .1 access MyROGroup any noauth 0 all none none and here is res_snmp.conf [general] subagent = yes enabled = yes Thanks all.! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.441 / Virus Database: 268.18.3/699 - Release Date: 23/02/2007 01:26 p.m. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.4/702 - Release Date: 25/02/2007 03:16 p.m. ___ --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] Registration problem
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5504 REGISTER Max-Forwards: 70 Contact: SPA922 sip:[EMAIL PROTECTED]:1025;expires=3600 User-Agent: Linksys/SPA942-4.1.12 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to x.x.x.x : 1025 (NAT) Transmitting (NAT) to x.x.x.x:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc;received=x.x.x.x From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5504 REGISTER User-Agent: incore-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce Call-ID: [EMAIL PROTECTED] CSeq: 5503 REGISTER User-Agent: incore-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479 Content-Length: 0 Why the phone can not register? The password and username are ok. Thanks ___ --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] how to config chanspy
How can I do to select the channel to spy ? thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ralph Liebessohn Enviado el: Miércoles, 18 de Octubre de 2006 09:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] how to config chanspy On 10/17/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiru Hi, It could be very simple, like: exten = 123,1,ChanSpy() ; Spy all channels or more accuracy: exten =124,1,ChanSpy(SIP) ; Spy all sip channels if I can help you more, let me know! -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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] Login user
Hi list! I have asterisk 1.2.12 installed and i need that the users can make a logon and logoff whit theirs phones on my asterisk pbx. Anybody know how can I do this ? Thanks in advance. ___ --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] voicemailmain errors on CLI
You have to leave a message in the voicemail, then listen it and the error will not apear again. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Doug Lytle Enviado el: Miércoles, 13 de Septiembre de 2006 08:45 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] voicemailmain errors on CLI Benjamin Jacob wrote: Hello ppl, I am getting the following errors when accessing voicemails Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Just as the error states, the directory Old doesn't exist. Check to see if it does. If it is there, check it's permissions, if not then create it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Problem with Tycho Voicemail
Hi list! Im using Tycho software to see my voicemail, y can see de detail from the message but i cant hear de message. Somebody use that software any time ? have you the same problem ? Thanks ___ --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: problems with wevbmail
I could fix it. The problem was permissions on the directory /var/spool/asterisk/voicemail. Thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Steven Enviado el: Miércoles, 23 de Agosto de 2006 08:01 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Re: problems with wevbmail Try running apache as the asterisk user instead of apache My assumption is that apache or your apache user does not have access to the voicemail folders. -- -- Steven http://www.glimasoutheast.org Sergio R. D'Ippolito [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without problems but i cant see the messages on any folder. Thanks, Sergio. ___ --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] NAT problems
Try changing the configuration on your PAP2 linksys, more precisly the part where is the NAT parameters, try changing the options from NO to YES. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de andrutto Enviado el: Miércoles, 23 de Agosto de 2006 03:41 p.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] NAT problems Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting 488 Not Acceptable Here and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. Cheers Andrutto -- Zostan Dziewczyna Lata! http://link.interia.pl/f1997 ___ --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] Strange SIP response
I had the same problem. The problem was another sip extensions whit the same ip. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rich Adamson Enviado el: Martes, 22 de Agosto de 2006 11:21 p.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Strange SIP response Diego Andres Asenjo G. wrote: Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. Check the sip device config and make sure Do Not Disturb (DND), Call Forwarding, etc, have not be set. ___ --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] problems with wevbmail
I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without problems but i cant see the messages on any folder. Thanks, Sergio. ___ --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
