search for: sipv2

...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 protocol sipv2 session target sip-server ! dial-peer voice 8427005 voip application app_tr...

...-strip direct-inward-dial port 1/0/0 ! dial-peer voice 20 pots description INBOUND CALLS PSTN BRI1 incoming called-number 2012345.. no digit-strip direct-inward-dial port 1/0/1 ! dial-peer voice 30 voip description INBOUND CALLS VOIP ASTERISK destination-pattern 2051860.. session protocol sipv2 session target ipv4:y.y.y.y:5060 session transport udp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 40 voip description OUTBOUND CALLS VOIP CARRIER destination-pattern .+ session protocol sipv2 session target ipv4:x.x.x.x:5060 session transport tcp dtmf-relay sip-notify...

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very bad. If more than one access the conference room it starts to blip real badly. Thots, ideas greatly appreciated. -- respectfull...

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

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

...all outgoing calls from asterisk via the cisco out to pstn, how do I do that ? in the cisco I have this: dial-peer voice 1 pots destination-pattern [0-9]T no digit-strip direct-inward-dial port 3/0:D ! dial-peer voice 50 voip destination-pattern [0-9] voice-class codec 1 session protocol sipv2 session target sip-server no vad dtmf-relay rtp-nte ! ------- But theese to dialplans seem to interrupt each other. When an incoming call from PSTN goes through this the pattern can be matched by the first, and then be routed ot on the PSTN again, creating a loop. How do I do this in the...

...130 qualify=yes context=from-ccme allow=all insecure=port,invite canreinvite=no ; [ccme-outbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme trustrpid=yes sendrpid=yes allow=all canreinvite=no dtmfmode=rfc2833 And, in CME: ----------------- dial-peer voice 200 voip session protocol sipv2 incoming called-number 2155551212 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 101 voip description softphones 4-N destination-pattern 4[0-9] monitor probe icmp-ping session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad...

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

...5060;branch=z9hG4bK40e0db6c..From: "4721677891" <sip:4721677891@mysipprovider >;tag=as11f66db2..To: <sip:callednumber@mysipprovider>..Contact: <si p:4721677891@myhost>..Call-ID: 5ae4b68378e5edf8081a79253ddf8aef@mysipprovider ..CSeq: 102 INVITE..User-Agent: Asterisk SIPv2 (http://www.ast erisk.org CVS-HEAD-11/27/04-15:07:27 ) ..Date: Sat, 27 Nov 2004 21:39:44 GM T..Diversion: ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Ty pe: application/sdp..Content-Length: 265....v=0..o=root 4687 4687 IN IP4 21 2.33.133.195..s=session..c=IN IP4 myhost..t=0 0...

...rate 2015232 no fair-queue ! interface Serial3 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial0:15 no ip address encapsulation ppp isdn switch-type primary-net5 no cdp enable voice-port 0:D ! ! ! dial-peer voice 10 voip destination-pattern .T session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 42 pots destination-pattern .T direct-inward-dial port 0:D ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:IP...

We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern 9999$ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also tried using rtp payload-type nte to adjust the nte port value to 101 versus 100. SIP Notify doesn't work, cisco-rtp doesn't work. I have tried every possible dtmfmode= option...

...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-activation no modem InOut transport inpu...

Friends, After having spent many years working with the Asterisk SIP channel driver and the SIPv2 protocol, I have finally realized that this is a dead end. It's getting nowhere and it's way too complicated to set up, run and support in working code. After realizing this, I started a new standardization project together with my friends in Canada, Simon and Marc, to develop a working so...

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

...ured on the cisco (below) I get : Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Which is correct, meaning the context declaration is not being respected. ------ dial-peer voice 101 voip destination-pattern 10. session protocol sipv2 session target ipv4:10.0.0.133 dtmf-relay rtp-nte codec g711ulaw no vad ------- My bad or something else ?? TIA, Nathan. Here is a sip debug for that peer: Sending to 10.0.9.1 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.0.9.1:19206...

...non-zero on 'SIP/-0811b4b8' sip*CLI> --------------------------- cisco#sh voice call 1/0/1 vtsp level 0 state = S_CONNECTvpm level 1 state = FXOLS_CONNECT vpm level 0 state = S_UP -------------------------- dial-peer voice 999 voip destination-pattern 8... session protocol sipv2 session target ipv4:10.1.1.1:5065 session transport udp codec g711ulaw no vad ! ------------------------------------ exten => 6917,1,Answer exten => 6917,2,Wait(1) exten => 6917,3,VoiceMail(u${EXTEN}) exten => 6917,4,Hangup Thanks. Ben -------------- next part -------------- An HTM...

Greetings, I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from Asterisk. I have been unable to identify through the docs how specifically this should be configured in Asterisk and have not been able to get things working through trial and error. I am sure I am missing something fairly obvious here but any guidance (or example cfgs) would be much appreciated.

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

...s duplex auto speed auto ! interface Serial3/0:23 no ip address dialer-group 1 isdn switch-type primary-5ess isdn incoming-voice voice no cdp enable ! voice-port 3/0:23 connection plar 1000 ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern 1000 session protocol sipv2 session target ipv4:1.2.3.4:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2 pots destination-pattern 9T port 3/0:23 ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:1.2.3.5 ! ----...

....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-pattern ....... port 0:1 ! sip-ua retry invite 3 retry cancel 2 sip-server ipv4:xx.xx.xx.xx:506...