search for: nte

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 vers...

...the time. > > NOTICE[15371]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support > incomplete. Turn off on client if possible > > > > > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = kiki ; Default for incoming calls > allow=alaw ; Allow codecs in order of preference > ;allow=ilbc > ;allow=all > > > [gw] > type=user > host=213.232.xxx.x...

...e class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 pri-group timeslots 1-31 description E1 Beta-Test interface Serial0 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial1 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial2 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial3 no ip address shutdown clock rate 2015232...

...e on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test setup (to avoid NAT complications, etc...) Here's the local static one, which works fine: sip.conf: ---------- [general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ; [ccme-inbound] type=friend host=10.5.7.130 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 sen...

.... When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay rtp-nte toggled in my dial peer.. Thanks, Billy +--------------------------------------------------+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-...

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.

...allow=all ;allow=ulaw allow=alaw debug message: File chan_sip.c, Line 5590 (sip_request): Asked to get a channel of unsupported format ULAW while capability is ALAW Why Asterisk doesn't use the SAME codec with outgoing & incoming calls ? In my AS5300, dtmf is configured as dtmf-relay rtp-nte perhaps I should try with h245-signal or h245-alphanumeric ? ALL ideas will be really appreciated ! Cheers, Areski

...it with the mailing list archive. But now I have this problem again with the class library. What should I put in library/class/R/zzz.R? Everything I put in that file leads to the same error message. I'm running 0.61.3 on Linux. > knn1(train, test, cl) Error in .C("VR_knn1", as.integer(ntr), as.integer(nte), as.integer(p), : C/Fortran function not in load table Execution halted -Egon -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help&quot...

..., 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 smartest and easiest way ? /Mike

Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about this. What I did get is th...

...r digital recordings. Our test setup has a second Asterisk server with a Digium quad-span card in it acting as a TDM-VoIP gateway. We are shooting for scalability, so the Asterisk server itself does no transcoding or DSP. We have noloaded all codecs except one and moved any of the resource-intensive activities to the gateway and the support servers. Our production setup will replace the Asterisk TDM-VoIP gateway with a Cisco AS5400HPX Universal Gateway. MCI has an AS5400 waiting for us at the D-Lab, and while they are familiar with most aspects of it, they lack any experience configu...

I've been using R for the last few weeks, but not with much success. If I try sourcing test.R from the 'class' package, I get this error: Error in .C("VR_knn1", as.integer(ntr), as.integer(nte), as.integer(p), : C/Fortran function not in load table I checked the class.so file, and sure enough VR_knn1 is there. Why can't R find it? By the way, I'm running RedHat Linux, and downloaded the .rpm from lib.stat.cmu.edu/R/CRAN/bin/i386-linux/RedHat-5.0/R-base-0.6...

I've been using R for the last few weeks, but not with much success. If I try sourcing test.R from the 'class' package, I get this error: Error in .C("VR_knn1", as.integer(ntr), as.integer(nte), as.integer(p), : C/Fortran function not in load table I checked the class.so file, and sure enough VR_knn1 is there. Why can't R find it? By the way, I'm running RedHat Linux, and downloaded the .rpm from lib.stat.cmu.edu/R/CRAN/bin/i386-linux/RedHat-5.0/R-base-0.6...

...ed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my phone. When I call that number again via the PSTN, and I am prompted to enter my extension number Asterisk never "hears" the dtmf tones. I have done some digging around, and my guess is that the issue relates to the codec being used messing up the tones. Am I on the right track? Is there a ideal way to handle this? what do others do? I have posted my sip.conf...

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 /

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco (below) I get : Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot find extension c...

...). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 description RCN PRI at SF7 ! interface FastEthernet1/0 no ip address 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 destinat...

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back t...

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

...ion files is: extensions.conf [fax-in] exten => 99999,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif) exten => 99999,n,Answer() exten => 99999,n,Wait(3) exten => 99999,n,ReceiveFax(${INCOMING_FAXFILE}) sip.conf [general] canreinvite=no t38pt_udptl=yes disallow=all allow=alaw context=fax-in The CISCO peer configuration: dial-peer voice 6 voip destination-pattern 88T session protocol sipv2 session target ipv4:10.100.0.51 session transport udp dtmf-relay rtp-nte codec g711alaw fax-relay ecm disable fax nsf 000000 fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback none...