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...;s shell I get that t is registering, but not authenticated .1. from show reg. Any ideas would be appreciated. Only passwords were removed. Thanks, Greg Oliver I have included my SIP.....cnf file for review.. # SIP Configuration Generic File (start) # Proxy Server proxy1_address: "sip.cistera.com" proxy2_address: "pbx-nwcorp.nationwide.net" proxy3_address: "192.168.117.4" proxy4_address: "192.168.117.4" proxy5_address: "192.168.117.4" proxy6_address: "192.168.117.4" # Line 1 Settings line1_name: "74678"...

Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Thanks, Ben Blakely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/a7e575cc/attachment.htm

...ms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. > > > > -----Original Message----- > From: Greg Oliver <greg.oliver at cistera.com> > Sent: Saturday, February 02, 2008 2:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall > > > > On Feb 2, 2008, at 2:11 PM,...

...ough the firewall? In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -----Original Message----- From: Greg Oliver <greg.oliver at cistera.com> Sent: Saturday, February 02, 2008 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall On Feb 2, 2008, at 2:11 PM, John Von Essen <john at quonix.net&g...

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI "binding" improved since then? 2) Is there any chance of a "real" API for Perl? Thanks much! -Ken -- This message has been scanned for viruses and dangerous

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

I've got problem with connecting asterisk to CCM. Our side has Asterisk system other side CCM , ehrn i dial a number on other side channles created , connections established but nothing happend , just silence , and after some time busy tone. Sides sending ad reciving g711 codec , but we need that sides send and recive g729 (we have licenses) , if in h323 conf i try to : disallow=all ,

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

OK - I can successfully make calls from SIp phone through an asterisk 323 channel to a Cisco Call Manager and out a MGCP controlled gateway. The problem is that if the call is not answered within ~5 seconds, * gives the message "no one is available to take your call" and disconnects the call. If I answer b4 the 5 seconds - everything is good. Anywhere I need to set to get around

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an