search for: outbond

Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070501/af78de7a/attachment.htm

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

...nd,30) exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:) exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:) exten => 101,n(lbl_default_0),Hangup() exten => 101,n(lbl_default_1),Dial(SIP/9046260700 at vitel-outbond,30) exten => 101,n,Goto(lbl_default_0) Cisco phone stuff from a Cisco 7960: SIPDefault.cnf image_version: P0S3-08-9-00 proxy1_address: neocipher.net ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: "100" phone_password: "cisco" ; Limited to 31 charact...

hello how can i only nat specific ports? 80,110,.... greets puchu

I am having a problem getting the following logic to work, in a macro. Basically, if the caller ID matches, set the outbond trunk to a Zap channel, otherwise use a SIP provider. exten => s,n,Set(TRUNK=${IF($[${CALLERIDNUM} = 1234567890]?Zap/g1:SIP/LDPROVIDER)}) ; use PRI instead of SIP That works. The TRUNK variable is set properly. But the SIP LD provider requires a prepended code, so I say if the TRUNK...

...92.168.X.X used for the routing between ISP router and itself. I also received a /32 public ip for the fw itself and i've added to the ethx:1 alias . Problem is that kernel always decide that (because of default gw being on the private ip 192.168.X.X) he has to use the 192.168.X.X ip address as outbond interface. So every packet leaving (so i'm talking about OUTPUT table and not about FORWARD nor nat table) the fw comes from a 192.168.X.X ip and so never comes back (which is normal). Question is : how can i "mangle" output packets to appear coming from public ip and not from 192.168...

...nd,30) exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:) exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:) exten => 101,n(lbl_default_0),Hangup() exten => 101,n(lbl_default_1),Dial(SIP/9046260705 at vitel-outbond,30) exten => 101,n,Goto(lbl_default_0) exten => 102,1,Dial(SIP/102,20) exten => 102,n,Hangup ;This automatically calls the right mailbox using the ${CALLERIDNUM} variable in the current context (var ${CONTEXT}). exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) include => inbo...

Hallo! Finally we have E1 PRI connected to our Asterisk box. Now I have one question. My internal extensions (_XXX) are SIP phones connected to Asterisk. Our telco routes some public numbers (_71602XX and others) to our Asterisk via E1. Some internal extensions can be reached from outside using public numbers (e.g. 7160234 -> 200), and some others cannot. Everyone can call outside

Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter