search for: kpml

...t Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an intertnet-draft for kpml: http://tools.ietf.org/id/draft-ietf-sipping-kpml-07.txt, but it seems to be very old - "Expires June 25, 2005". I know that using MTP in SIP Trunk at CCM5 makes DTMF work in RFC2833, but MTP resource is ver...

...es, etc. The following is simply an FYI from the presentation, FWIW. Existing IP Phones (7905/7912/7940/7960): Transport: UDP Call Signalling: rfc2833 Security: Digest Auth Feature parity to SCCP: None Sip Enhanced Phones (7911/7941/7961/7971/7970): Transport: UDP/TCP/TLS Call Setup: KPML and Dial Rules (KPML sends each dialed digit in a packet) Call Signaling: KPML, rfc2833 Security: CAPF/CTL/TLS Feature parity to SCCP: Almost the same (will be later) Notes: 1. All cisco phones with dual rj45's pass BPU packets (spanning tree) through internal switch ports. The swit...

I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC 2833? ----------------------------------------- Disclaimer: Thi...

...lt;/preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <phoneLabel>Test2</phoneLabel> <stutterMsgWaiting>2</stutterMsgWaiting> <callStats>false</callStats> <offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCal...

...rt=UDP> From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.7797 Allow-Events: presence, kpml Content-Length: 0 <-------------> [Oct 7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct 7 13:46:52] --- (13 headers 0 lines) --- [Oct 7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct 7 13:46:52] Sending to 192.168.114.20 : 5060 (no NAT) [Oct 7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct 7 13:46...

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

...2.168.1.61 Max-Forwards: 70 Date: Thu, 26 Feb 2015 05:52:42 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7975G/8.5.3 Contact: <sip:111 at 192.168.1.61:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Allow-Events: kpml,dialog Content-Length: 322 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 626 0 IN IP4 192.168.1.61 s=SIP Call t=0 0 m=audio 30354 RTP/AVP 0 8 18 116 101 c=IN IP4 192.168.1.61 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18...

...fAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>false</natEnabled> <natAddress></natAddress> <phoneLabel>Alker Study</phoneLabel> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>true</callStats> <silentPeriodBetweenCal...

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

...YwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 AST.ER.ISK.IP s=Z c=IN IP4 AST.ER.ISK.IP t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Invite that Asterisk sends: PU.BL.IC.IP:5070 &gt...

...IyNzk. > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, > INFO, SUBSCRIBE > Content-Type: application/sdp > Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri > User-Agent: Z 3.2.21357 r21367 > Allow-Events: presence, kpml > Content-Length: 239 > > v=0 > o=Z 0 0 IN IP4 AST.ER.ISK.IP > s=Z > c=IN IP4 AST.ER.ISK.IP > t=0 0 > m=audio 8000 RTP/AVP 3 110 8 0 98 101 > a=rtpmap:110 speex/8000 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0...

...ber: 1 Method: REGISTER Expires: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 On Mon, Feb 15, 2016 at 2:53 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > >> Thanks for the mighty quick response, Joshua! >> >> I am using Zoiper on Linux softclient: >> REGISTER sip:<ipAddr>;transp...

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

...NS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1 Allow-Events: kpml,dialog Content-Length: 685 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71 s=SIP Call t=0 0 m=audio 10032 RTP/AVP 0 8 18 102 9 116 101 c=IN IP4 10.168.154.71 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000...

...CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 2.2.2.2 s=Z c=IN IP4 2.2.2.2 t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000...

...igest username="demo-alice",realm="asterisk",nonce="[removed]",uri=" sip:6002 at 192.168.1.139 ;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]" Allow-Events: presence, kpml Content-Length: 245 v=0 o=Z 0 0 IN IP4 146.115.163.234 s=Z c=IN IP4 146.115.163.234 t=0 0 m=audio 8000 RTP/AVP 0 3 110 8 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <--- Transmitting SIP response (319 byt...

....samba.org.AVP; Wed, 15 Aug 2001 23:47:17 +0400 (MSD) Received: from homepc (kfml.dialup.kirov.ru [194.84.203.239]) by proxy.kirov.ru (8.11.2/8.11.2) with ESMTP id f7FJlFS77097 for <samba@lists.samba.org>; Wed, 15 Aug 2001 23:47:16 +0400 (MSD) Date: Wed, 15 Aug 2001 23:45:46 +0400 From: KPML <fml@ezmail.ru> X-Mailer: The Bat! (v1.53bis) Reply-To: KPML <fml@ezmail.ru> Organization: KPML Message-ID: <18927410614.20010815234546@ezmail.ru> To: samba@lists.samba.org Subject: Samba block files..., How to make... MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii C...