search for: progress_ind

...t 2/1 ! voice-port 2/2 ! voice-port 2/3 ! voice-port 2/4 ! voice-port 2/5 ! voice-port 2/6 ! voice-port 2/7 ! ! ! dial-peer voice 1 pots destination-pattern .T fax rate voice direct-inward-dial port 1/0:23 ! dial-peer voice 2017005 voip application app_transfer destination-pattern 5132017005 progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 1 session protocol sipv2 session target sip-server ! dial-peer voice 2017006 voip application app_transfer destination-pattern 5132017006 progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 1 session...

...onnection plar opx 1001 station-id number 100 caller-id enable ! voice-port 1/0/1 input gain 10 output attenuation 10 no comfort-noise caller-id enable ! voice-port 1/1/0 ! voice-port 1/1/1 ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 100 pots destination-pattern .T progress_ind setup enable 3 progress_ind progress enable 8 port 1/0/0 ! dial-peer voice 2 voip destination-pattern 1... progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 1 session protocol sipv2 session target ipv4:xxx.xxx.xxx.xxx:5060 session transport udp dtmf-relay h245-a...

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

...l1 no ip address shutdown ! interface FR-ATM20 no ip address shutdown ! ip default-gateway xx.xx.xx.xx ip classless ip route 0.0.0.0 0.0.0.0 xx.xx.xx.xx no ip http server ! ! ! snmp-server community xxxxxx RO ! voice-port 0:1 ! voice-port 1:1 ! ! ! dial-peer voice 1 voip destination-pattern T progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 10 session protocol sipv2 session target ipv4:xx.xx.xx.xx session transport udp dtmf-relay rtp-nte no vad ! dial-peer voice 110 pots incoming called-number .... direct-inward-dial ! dial-peer voice 100 pots destination-patter...

...levant Config ---------------------------------------------------------------------------- --- ... ! voice class codec 1010 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g723ar63 codec preference 4 g723r63 ! ... ! dial-peer voice 1010 voip destination-pattern 85.. progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 1010 session protocol sipv2 session target ipv4:xxx.xxx.xxx.xxx (ASterisk Box) dtmf-relay cisco-rtp rtp-nte h245-signal h245-alphanumeric ! ---------------------------------------------------------------------------- ---

...T1 to CLEC no ip address load-interval 30 isdn switch-type primary-ni isdn incoming-voice modem no cdp enable voice-port 3/0:D bearer-cap Speech dial-peer voice 1 pots incoming called-number 21255512[00-50] direct-inward-dial ! dial-peer voice 100 voip destination-pattern 21255512[00-50] progress_ind setup enable 3 session protocol sipv2 session target ipv4:10.10.10.10 codec g711ulaw no vad ! dial-peer voice 1000 pots destination-pattern .......... port 3/0:D sip-ua retry invite 4 retry response 3 retry bye 2 retry cancel 2 sip-server ipv4:10.10.10.10 line 1/00 1/59 no flush-at-act...

Hey guys, I am trying to pass the parameters for the file being upload, but the params[] are not being passed. Ive read through some posts and realize this is an issue with RoR. I read through some of the posts and someone suggested using: http://svn.kylemaxwell.com/form_remote_upload/trunk/ I tried to install it, and although I thought I installed it correctly, it doesn''t work.

...ination-pattern 1000 port 2/1/0 authentication username 1000 password <password> ! dial-peer voice 1001 pots description Binds to FXS Port 2/1/1 destination-pattern 1001 port 2/1/1 authentication username 1001 password <password> ! dial-peer voice 200 voip destination-pattern .T progress_ind progress enable 8 session protocol sipv2 session target ipv4:<ip of asterisk server>:5060 session transport udp codec g711ulaw sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 registrar ipv4:<ip of asterisk server> expires 60 sip-server i...

Hi I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then

Hello, We have a DID partner sending traffic to Asterisk via SIP, but we are not hearing ringtones. When we call the same extension via SIP, we can hear that's it"s ringing (virtually).. Is is something related with call-progress not recognized by DID provider ? Thanks, ________________________ a b d o u l aba at gcomnetworks.com SIP: (131) 229-1002 at sip.freeipcall.com

Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway