Using MJ dongleless with Csico UC520 as a SIP trunk

[gumby357]

Just purchased an MJ for a Home Lab and have a few questions foir the Gurus on this board. I have read through many post and See that it works and may not work so looking for clarification.

1. I see that in the past you can set up a SIP connection if I get PDUMP. Dose the PMDUMP still work with the newest Versions and if not who can I get the SIP info ?

2.Do you have to change anything on the SIP connection when they rquires a software update if not using the dongle at all.

3. Most folks are using SIP with a diffrent softphone. I am planing on setting up a SIP connection with a Dial peer on a Cisco UC520. Will it work with out the MJ software at all as long as I have my SIP creds ?

Thanks In adavnce.

[gumby357]

Pmdump Still works Just have to get the PMDUMP command in there as soon as the Magicjack.exe pops up in the list of proccesses when you plug in the device.

now on to trying this as a SIP dial peer with a cisco UC520

[gumby357]

I have my UC520 working with the Magic Jack service without needing the dongle.
I think this set up will work with a any Cisco Voice Gateway or CME set up.
Looks like I can have 4 calls at once on the sip trunk (Have not tested beond that).

Here are the pertinent parts of the config

! USED ON OUTBOUD DIALPEERS TO CORECT CALLING ID TO
!MATCH WHAT MAGIC JACK SHOULD BE
!
voice translation-rule 1111
rule 1 /.*/ /EXXXXXXXXXX01/
!
!USED ON INBOUND DIALPEER TO SEND CALL TO IN FORM MJACK
!AUTO aTTENDANT SCRIPTSERVICE
voice translation-rule 2001
rule 1 /EXXXXXXXXXX01/ /XXX/
!
!SIP LOCALHOST SECTION WILL SEND OUT THE SIP REQUEST IN THE FORMAT THEY ARE EXPEXTING ([email protected])
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
subscription maximum originate 10
localhost dns:talk4free.com
no update-callerid
!
!USED IN FIREWALL RULES TO ALLOW MJACK BACKIN
access-list 104 permit udp host proxy1.cleveland.talk4free.com eq 5070 any
!
!USED TO ALLOW UDP VOICE STREAMSBACKINTROUH THE FIREWALL
access-list 104 permit udp any any range 16384 32767
!
!OUTBOUND/INBOUND DIALPEER FOR OUTBOUND CALLS THIS IS JUST THE 10DIGIT LOCAL DIALPEER
!YOUWILLNEED OTHERS FOR LD AND 411,911 ECT
dial-peer voice 1000 voip
translation-profile incoming AA_Profile
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
destination-pattern ..........
session protocol sipv2
session target sip-server
incoming called-number .
dtmf-relay rtp-nte
codec g729r8
no vad
!
!SIP CREDENTALS FROM PMDUMP OR OTHER MAGICJACK CRACKING TOOLS
!
sip-ua
credentials username EXXXXXXXXXX01 password (YOUR 20 DIGIT PSWD) realm proxy1.cleveland.talk4free.com
authentication username EXXXXXXXXXX01 password 7 (YOUR 20 DIGIT PSWD)
retry invite 2
retry response 3
retry bye 3
retry prack 6
retry options 10
timers expires 300000
registrar dns:proxy1.cleveland.talk4free.com:5070 expires 840 (server nearest you )
sip-server :proxy1.cleveland.talk4free.com:5070
notify telephone-event max-duration 500
!
This should work on any Cisco router with voice software
have fun.

[mberlant]

registrar ipv4:67.88.10.198:5070 expires 3600 (server near me)
registrar ipv4:66.104.96.198:5070 expires 3600 secondary (server near me)

I recommend changing this timer from 3600 (one hour) to no more than 840 (14 minutes). In the very likely case that your Cisco can't interpret the 870 value that is supposed to come back in the Register OK packet, this will ensure that your Cisco reregisters before the MJ server notices that it has not. With a 3600 value you run the risk of not being registered for the latter 45 1/2 minutes of every hour of operation.

Thank you for this great contribution.

[gumby357]

registrar ipv4:67.88.10.198:5070 expires 3600 (server near me)
registrar ipv4:66.104.96.198:5070 expires 3600 secondary (server near me)

I recommend changing this timer from 3600 (one hour) to no more than 840 (14 minutes). In the very likely case that your Cisco can't interpret the 870 value that is supposed to come back in the Register OK packet, this will ensure that your Cisco reregisters before the MJ server notices that it has not. With a 3600 value you run the risk of not being registered for the latter 45 1/2 minutes of every hour of operation.

Thank you for this great contribution.

Thanks will give that a shot.
I have not noticed any disruption in registration as of yet but want it to look as much like the MJ device as possible.
Thanks !!
Also for now I have taken the Secondary registrar out of the setings.

[mberlant]

You won't notice the disconnection, but your callers will.

During that 45 minute window each hour your Cisco will think it's Registered, but MJ will not think so. Your Cisco will continue to give you Dial Tone when you ask for it, and will automatically re-Register and retry the call when it fails during this window. You won't notice the 100-300ms it takes for this extra step to occur.

Anybody who calls you during this window, though, will be dumped directly into your MJ voice mail, because MJ is under the belief that you are off the air.

[gumby357]

You won't notice the disconnection, but your callers will.

During that 45 minute window each hour your Cisco will think it's Registered, but MJ will not think so. Your Cisco will continue to give you Dial Tone when you ask for it, and will automatically re-Register and retry the call when it fails during this window. You won't notice the 100-300ms it takes for this extra step to occur.

Anybody who calls you during this window, though, will be dumped directly into your MJ voice mail, because MJ is under the belief that you are off the air.

Thanks fo rthe heads up have implemented the change.
Also goign to try changing the codec to G.729

[gumby357]

Working with G729 !! with no VAD because I hate VAD probably will work with it on the cisco side but VAD =BAD
UC520#sho sip call
Call 1
SIP Call ID : [email protected]
State of the call : STATE_RECD_PROCEEDING (4)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : Exxxxxxxxxx1
Called Number : 1xxxxxxxxxx
Bit Flags : 0xC04018 0x100 0x4200
CC Call ID : 1159
Source IP Address (Sig ): X.X.X.X
Destn SIP Req Addr:Port : 67.88.10.198:5070
Destn SIP Resp Addr:Port: 67.88.10.198:5070
Destination Name : proxy1.cleveland.talk4free.com
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 1159
Stream Type : voice+dtmf (1)
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
Media Source IP Addr:Port: X.X.X.X:17812
Media Dest IP Addr:Port : 67.88.10.203:20776
Orig Media Dest IP Addr:Port : 0.0.0.0:0

[mberlant]

Don't expect for MJ to perform any transcoding in their proxy servers, nor to pay the licensing fee on G.729, at the price they are charging us. Your success indicates that, for the number you called, the termination provider supports G.729 at their outbound gateway. You may observe that calls to other destinations (especially other countries) will only connect with G.711, though more and more termination providers are supporting G.729 (as well as GSM and G.723 or G.726).

Enjoy.

[gumby357]

Don't expect for MJ to perform any transcoding in their proxy servers, nor to pay the licensing fee on G.729, at the price they are charging us. Your success indicates that, for the number you called, the termination provider supports G.729 at their outbound gateway. You may observe that calls to other destinations (especially other countries) will only connect with G.711, though more and more termination providers are supporting G.729 (as well as GSM and G.723 or G.726).

Enjoy.

Under stand, But is not the G.729 codec a public domain codec ?

That is why Cisco and many other vendors use this codec in their VOIP soluttions does not cost them anything.

[mberlant]

No, G.729 is most definitely a patent protected CODEC. The patent holder has engaged the services of Sipro Lab, in Montreal, Quebec, to administer its licensing. The licensing structure is such that the royalty fee can be incorporated into a piece of hardware easier and more inexpensively than into a piece of software, most notably because the software module must be hard linked to a specific ethernet interface device.

As an example, in the Asterisk world Digium has struck a deal with the owners of G.729 to resell licenses to individual Asterisk users at the bargain basement price of $10 per instance. That's $10 times the number of simultaneous G.729 transcodings you care to support. Needless to say, that can run into serious money very quickly.

[gumby357]

Don't expect for MJ to perform any transcoding in their proxy servers, nor to pay the licensing fee on G.729, at the price they are charging us. Your success indicates that, for the number you called, the termination provider supports G.729 at their outbound gateway. You may observe that calls to other destinations (especially other countries) will only connect with G.711, though more and more termination providers are supporting G.729 (as well as GSM and G.723 or G.726).

Enjoy.

I had a problem with 800 tool free numbers with the g729 codedc
I had g729 as the first preferance and the G711 as the second but it would not negotiate to the second pref.
I even changed the pref arround and still would fail.
I had to set up an 8XX toll fre dial-peer and specify G711 to get it to work.

I would get the following error message in the sip debugs on any toll free calls.

"202903: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Info/sipSPIDoAudioNegot
iation: No matching voice codec found for m-line 1
202904: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Info/sipSPIDoDTMFRelayN
egotiation: m-line index 1
202905: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Info/sipSPICheckDynPayl
oadUse: Dynamic payload(101) could not be reserved.
202906: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Info/sipSPIDoDTMFRelayN
egotiation: RTP-NTE DTMF relay option
202907: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Info/sipSPIDoDTMFRelayN
egotiation: Case of partial named event(NE) match in fmtp list of events.
202908: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Info/sipSPIGetSDPDirect
ionAttribute: No direction attribute present or multiple direction attributes th
at can't be handled for m-line:1 and num-a-lines:0
202909: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Error/sipSPIDoAudioNego
tiation: Media negotiation failed for m-line 1
202910: Feb 14 11:14:36.899 EST: //4849/6398AEE68646/SIP/Error/sipSPIDoMediaNego
tiation:
no valid fax or audio streams"

[gumby357]

No, G.729 is most definitely a patent protected CODEC. The patent holder has engaged the services of Sipro Lab, in Montreal, Quebec, to administer its licensing. The licensing structure is such that the royalty fee can be incorporated into a piece of hardware easier and more inexpensively than into a piece of software, most notably because the software module must be hard linked to a specific ethernet interface device.

As an example, in the Asterisk world Digium has struck a deal with the owners of G.729 to resell licenses to individual Asterisk users at the bargain basement price of $10 per instance. That's $10 times the number of simultaneous G.729 transcodings you care to support. Needless to say, that can run into serious money very quickly.

Wow Have been installing Nortell, 3com and Cisoc Voip solutions for 10 years using the vendor provided G.729 codec. I know we had to license others that is why we used G729. Unless the vendors have licensed it already for use on their respective system ?


Thanks

[mberlant]

Wow Have been installing Nortell, 3com and Cisoc Voip solutions for 10 years using the vendor provided G.729 codec. I know we had to license others that is why we used G729. Unless the vendors have licensed it already for use on their respective system ?

Yes, indeed. It is rather painless and inexpensive to incorporate G.729 into a production run of hardware of the size that Cisco or NorTel or Linksys/Sipura crank out. This is partially because in a hardware solution there is no license management necessary -- it is impossible for a hardware interface to handle more than the number of simultaneous conversations than it is licensed for, and they buy bulk licenses for the entire production. That is why so many hardware manufacturers include G.729.

[gumby357]

Wow Have been installing Nortell, 3com and Cisoc Voip solutions for 10 years using the vendor provided G.729 codec. I know we had to license others that is why we used G729. Unless the vendors have licensed it already for use on their respective system ?

Yes, indeed. It is rather painless and inexpensive to incorporate G.729 into a production run of hardware of the size that Cisco or NorTel or Linksys/Sipura crank out. This is partially because in a hardware solution there is no license management necessary -- it is impossible for a hardware interface to handle more than the number of simultaneous conversations than it is licensed for, and they buy bulk licenses for the entire production. That is why so many hardware manufacturers include G.729.

Thnks for the info

[mattvirus]

Fantastic thread! Just stumbled on this after picking up my magicjack and getting it working (analog-style) with my UC520.

[iox]

I did the same steps as descrive. I'm getting Bad Gateway from Majic Jack server proxy01.miami.talk4free.com.

Do I need MJMD5? or is this still working?

[iox]

gumby357,

Can you upload the translation profile of you setup I think I'm missing that