Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO
El Sat, Jan 19 de 2008 a las 23:35 -0600, Moises Silva comentaba: First, let me say I am confused about this: I've changed the line (chan_unicall.c): uc_callparm_calling_party_category(callparms, UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL); to uc_callparm_calling_party_category(callparms, UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL); because without this I cant receive calls from the telco. With or without this I can't place calls to the pbx. I am quite sure you have made a mistake in this statement, why? simply because this code is executed when YOU START the call to the far end (whatever it is, Telmex or the other PBX), so it makes no sense to say that w/o that change you can't receive calls, no sense at all. I am sure you messed up somewhere else in the configuration files just like possibly you are doing right now for the PBX. Hmmm.. I was pretty sure that if I dont change that line I cant receive calls from telmex.. but let try it againg and I will tell you what it was. In anycase, I am about to make a new release of chan_unicall Asterisk driver that will include a way to modify the calling party category from the dialplan extensions.conf Wouldn't it be better if that could be done in unicall.conf? As with the other options like protocolvariant and protocolend ?? Anyway.. thanks for doing that update ;) I would be glad to know when it is available. Now, regarding your problem when receiving calls from the pbx, I think you have configured the PBX to not send ANI digits, and you configured chan_unicall to expect ANI digits, hence the timeout. Try configuring Asterisk with 0 callerid for the PBX side, or configure the other PBX to send the proper number of ANI digits. Well.. in the first place.. that pbx is not mine, I didnt configured it and I cant even touch it, Im just putting asterisk in between right now. Im gonna try that.. Thanks for the help! -- Regards.. Victor Toofic ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R2-Unicall Asterisk as CPE and as CO
UniCall/65 - 5 off [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:42 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 off - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 5 on [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 on - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 5 off [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 off - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 0 on [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 on - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 0 off [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:43 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 off - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:44 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 0 on [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:44 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 on - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:44 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 0 off [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:44 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1 off - [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:45 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 7 on [2/DETECTED/Group A /DNIS request ] Jan 16 12:20:45 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 6 on - [2/DETECTED/Group C /Category req ] Jan 16 12:20:45 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 - 7 off [2/DETECTED/Group C /Category req ] Jan 16 12:20:45 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 6 off - [2/DETECTED/Group C /Category req ] Jan 16 12:20:50 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 R2 prot. err. [2/DETECTED/Group C /Category req ] cause 32771 - T3 timed out Jan 16 12:20:50 DEBUG[4085] chan_unicall.c: MFC/R2 UniCall/65 1001 - [1/IDLE/Idle /Idle ] Jan 16 12:20:50 NOTICE[4085] chan_unicall.c: Unicall/65 event Protocol failure I've tried a lot of variations with these variables: protocolvariant=mx,10,8 group=3 immediate=yes usecallerid=yes protocolend=co adding 8 16 24 at the end of protocolvariant (4th param), using immediate=no, usecallerid=no, protocolend=cpe and protocolend=co.. I mean.. I've tried a lot of combinations without success. Dont know if this is a configuration related problem.. or there is something else I am missing ?? Hope someone could help me. Thanks... PS. Sorry for the long mail. -- Greetings.. Victor Toofic ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticate by IP?
El Mon, Oct 22 de 2007 a las 15:59 -0500, Carlos Chavez comentaba: Hola José. Gracias por tu contestación. Lo que me estas especificando el para hacer llamadas de salida (PEER). Yo necesito autentificar a un usuario de entrada, voy a intentar haciendo algo parecido solo cambiando a type=user para ver si así funciona. type=peer also works for incoming calls. In this case (peer) asterisk only checks the IP the call is coming from and uses the context you defined there. If you use type=user you will need to specify a username and a secret. -- Greetings.. Víctor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
El Fri, Aug 03 de 2007 a las 20:24 +0500, Rizwan Hisham comentaba: I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why one thing i still dont understnd. if the device we are using is a computer, and we r running a softphone on it. and side by side we are also surfing the net. then why is it so that web content is coming into the computer without any problem but rtp data is not. i think both the web application and softphone are using computer's local ip address in their requests. So whats the reason for this? Simple, web content uses TCP while RTP uses UDP to carry the data. In TCP your computer needs to establish a connection with the remote side before each one can send any data, the device which is doing the NAT realizes that and creates a bridge between your computer's IP/Port and the remote site IP/Port. In the case of SIP/RTP over UDP is different. Your softphone sends the signaling over a UDP port, the remote site receives the data and responses back to the IP/Port it recived it from (the IP/Port of the NATing device), the device which is doing the NAT knows that you have recently send data over that IP/Port and routes it back to you. Thats why SIP signaling can work fine even behind a NAT (nat=yes). RTP flow is also different. Your softphone specify it wants to receive RTP in a IP/Port (private IP/Port), when the remote site wants to send you RTP data it cannot be routed because that address is private, it cannot send the data to the address of the NATing device because the port this device is using for your outgoing RTP is different than the port you specified. So the RTP that is destinated to you gets lost. I understand how stun works but thanx for explaining it in so simple and concise way. One other question which has been bothering me is: If the client phone is behind nat, that means there is NATTING going on between public internet and local net. Then why do we need stun? NATTING should handle the problem itself as it does for other applications running on the same computer where softphone is also running. NATting can, in someway, handle the problem when you originate the call, but it cannot do it when someone wants to reach you later. The SIP header Contact is used for this, when someone wants to reach you it uses the address you specified in that header, so it must be a public IP address which you obtained from the STUN server or another mean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/Dont know how to handle Accepted
El Sun, Jul 29 de 2007 a las 20:04 +0800, Steve Underwood comentaba: What versions of software did you use to get a screwed up result like that? The message Don't know how to handle signalling event Accepted is printed at the end of a case statement which does handle that event. I the publicly available versions of unicall, and can't see how that could go wrong, even if you mix components from different versions. Now I can see what was my mistake. I was using the libraries: http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/ but there's no chan_unicall.c in there, so I took it from: http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/ and miss-patched the call events enum in unicall.h. I was using that mixture because I got some errors trying to compile unicall-0.0.3pre11. Now I solved the compile issue in unicall-0.0.3pre11 and Im using that, I can't still get it to work but I think it's a miss-configuration in some of the endpoints. I'll keep trying. Thnks.. -- Regards, Víctor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall/Dont know how to handle Accepted
UniCall/3 - 9 on [2/DETECTED/Group C /ANI request] MFC/R2 UniCall/3 1 on - [2/DETECTED/Group C /ANI request] MFC/R2 UniCall/3 - 9 off [2/DETECTED/Group C /ANI request] MFC/R2 UniCall/3 1 off - [2/DETECTED/Group C /ANI request] MFC/R2 UniCall/3 - 9 on [2/DETECTED/Group C /ANI request] MFC/R2 UniCall/3 3 on - [2/DETECTED/Group B /Go to grp II] MFC/R2 UniCall/3 - 9 off [2/DETECTED/Group B /Go to grp II] MFC/R2 UniCall/3 3 off - [2/DETECTED/Group B /Go to grp II] MFC/R2 UniCall/3 - 2 on [2/DETECTED/Group B /Go to grp II] Unicall/3 event Offered CRN 32769 - Offered on channel 0 (ANI: 814777, DNIS: 83329276, Cat: 1) MFC/R2 UniCall/3 Call control(5) MFC/R2 UniCall/3 Accept call MFC/R2 UniCall/3 1 on - [2/OFFERED /Group B /Accepted Paid] MFC/R2 UniCall/3 - 2 off [2/OFFERED /Group B /Accepted Paid] MFC/R2 UniCall/3 1 off - [2/OFFERED /Group B /Accepted Paid] MFC/R2 UniCall/3 Answer guard expired Unicall/3 event Accepted Unicall/3 Don't know how to handle signalling event Accepted the log stops here and there's only ringing in the phone. These are my configurations: zaptel.conf: span=1,0,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 loadzone=mx defaultzone=mx unicall.conf [channels] context=incoming usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no loglevel=255 protocolclass=mfcr2 protocolvariant=mx,10,10 protocolend=cpe group=2 channel = 1-15 channel = 17-31 I hope someone can point out my mistakes. Thanks... PS. I know my Asterisk's version is a bit old, but I cant upgrade right now because it's been used by some customers. -- Victor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/Dont know how to handle Accepted
El Fri, Jul 27 de 2007 a las 11:18 -0500, Carlos Chavez comentaba: Try changing protocol variant to: protocolvariant=mx,10,4 I've already tried several combinations. I dont know exactly what means each number but I could get an idea according what I saw in the logs. Even so, I've tried it and the only difference is: CRN 32769 - Offered on channel 0 (ANI: 814777, DNIS: 8332, Cat: 1) Thats why I put 10,10.. because I wanted the ANI to be 10 digits length and the DNIS 10 digits length. Usually only you only need the last 4 digits of the DID to identify it. Also, you only get 8 digits for the DNIS because that is the maximum anyone will dial to get to you (7 if you are in a smaller city). ANI is always 10 in Mexico. I'm not connecting to any telco, instead I'm connecting the E1 to a CISCO device which is supposedly to be correctly configured. Said that, I am realizing that maybe I should put some other value instead of 'mx'. I will give it a try. If this is correct.. Do you know what should be there? Thanks.. -- Victor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/Dont know how to handle Accepted
Ok, my boss is telling me that Im using Category 1 in the signaling and he is asking me to change it to Category 2. R2 Incoming Voice(0/0): DSX (E1 0/0:0): STATE: R2_IN_CATEGORY R2 Got Event R2_TONE_OFF Enter r2_comp_category r2_reg_generate_digits(0/0:1(1)): Tx digit '#' htsp_digit_ready_up(0/0:1(1)): Rx digit='1' We are connecting our Asterisk to a Gateway AS5400. So, Im wondering how am I supposed to do that change? I cant see any place to configure that. What should I do? Thanks... -- Victor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/Dont know how to handle Accepted
El Fri, Jul 27 de 2007 a las 16:07 -0500, Victor Toofic comentaba: Ok, my boss is telling me that Im using Category 1 in the signaling and he is asking me to change it to Category 2. R2 Incoming Voice(0/0): DSX (E1 0/0:0): STATE: R2_IN_CATEGORY R2 Got Event R2_TONE_OFF Enter r2_comp_category r2_reg_generate_digits(0/0:1(1)): Tx digit '#' htsp_digit_ready_up(0/0:1(1)): Rx digit='1' We are connecting our Asterisk to a Gateway AS5400. So, Im wondering how am I supposed to do that change? I cant see any place to configure that. What should I do? Sorry for asking something that had been answered previously. I've found on this list that it is hardcoded in chan_unicall.c and its necessary to change: uc_callparm_calling_party_category(callparms, UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL); with uc_callparm_calling_party_category(callparms, UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL); Now my boss is partly happy ;) as we are facing some other problems.. I will invesitgate it later. Thanks... PS. Would be worth anough adding a configuration parameter in unicall.conf to specify the CallerCategory? -- Victor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...
El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba: I have a customer that is complaining that any call coming in from Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel 1.4.3 and all the MFC/R2 patches and libraries. All other calls go out and come in, just Nextel seems to have this problem. The phone company technician connected a PBX emulator on the line and that one could receive the calls from Nextel. Would you please tell me what version of the libraries are you using, Im trying to get running * with R2 without success. Thanks... Víctor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
El Thu, Jun 28 de 2007 a las 20:18 +0530, [EMAIL PROTECTED] comentaba:
Hi,
I am trying to establish call through sip phone between two PC
connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Now, I am tying to dial from 1st PC to 2nd PC
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1st PC is calling directly to 2nd PC not through
asterisk server
I am doing the following additions in configuration files
1) sip.conf
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
[phone1]
type=friend
host=192.168.1.149
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
[phone2]
type=friend
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)
Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
I guess thats why the phones are talking directly: [EMAIL PROTECTED]
Either call extension '11' from phone1 or add a extension named 'phone2' to
extensions.conf and call that extension ('phone2') without the ip address.
Make sure your softphones are correctly configured: sip proxy address (*
address), username, etc.
Btw, both devices in sip.conf are declared as 'friend'.. thus you must specify
a secret (and optionally a username):
[phone2]
type=friend
username=phone2
secret=qwerty
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
It is not being called through asterisk server running on linux m/c. It
is calling directly. As, I am running sip debub but no packet dumping is
taking place. Can anybody will tell me the error I am doing.
Thanx and regards
sanchal
___
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Re: [asterisk-users] TCP-UDP SIP proxy?
El Wed, Jun 06 de 2007 a las 19:14 +0300, Yehavi Bourvine +972-8-9489444 comentaba: Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? OpenSER? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Testing
El Thu, Apr 26 de 2007 a las 13:45 -0400, David Ruggles comentaba: Does anyone have any scripts or templates that could be used by an Asterisk box to call an IVR for stress testing? This task seems like it would be ideally suited to Asterisk and I wanted to see if anyone had already done this before I started trying to roll my own. Yes, you can use the SIPp tool with pcap support enabled: http://sipp.sourceforge.net/doc/reference.html#UAC+with+media You can take that template and easily add something like this: pause milliseconds=13000/ nop action exec play_pcap_audio=pcap/dtmf_2833_2.pcap/ /action /nop pause milliseconds=5000/ nop action exec play_pcap_audio=pcap/dtmf_2833_1.pcap/ /action /nop pause milliseconds=700/ nop action exec play_pcap_audio=pcap/dtmf_2833_2.pcap/ /action /nop pause milliseconds=700/ Hope this can help! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming audio file while working in background ?
El lun, ene 22 de 2007 a las 16:58 -0700, Darren Nay comentaba: Ideally I would like to be able to play an audio file to the caller while making outbound calls in the background (via the Dial app) and then discontinue the audio file stream and bridge the calls once an outbound call is connected. Maybe you can execute a macro when the called party answers and still provide music on hold or something to the caller, this is done with the opcion 'M' of the Dial app. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Look at the Example Nr 2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba: however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting native bridge of
El vie, nov 17 de 2006 a las 04:34 +0530, Vicky comentaba: Thats really strange .. if you have made canreinvite=no then it should not even attampt native bridging and should transcode codecs ..something's fishy here .. Also try to put canreinvite=no in testulaw exntension too . So why do I have audio in both ways using 2 IP Phones with 2 different codecs and getting 'Attempting native bridging' at the same time? I've always had canreinvite=no in all my extensions. This is my sip.conf: [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testulaw] type=friend host=dynamic username=testulaw context=astertest canreinvite=no disallow=all allow=ulaw [testalaw] type=friend host=dynamic username=testalaw context=astertest canreinvite=no disallow=all allow=alaw [testg723] type=friend host=dynamic username=testg723 context=astertest canreinvite=no disallow=all allow=g723 [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 [1001] type=friend host=dynamic username=1001 context=astertest canreinvite=no disallow=all allow=g729 [1010] type=friend host=dynamic username=1010 context=astertest canreinvite=no disallow=all allow=ulaw Thanks again! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting native bridge of
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba: g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? I get the same message even if I'm not using g729: --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80 ulawgsm UAS --- * --- UAC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting native bridge of
El jue, nov 16 de 2006 a las 11:35 -0600, Victor Toofic comentaba: El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba: g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? I get the same message even if I'm not using g729: --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80 ulawgsm UAS --- * --- UAC Ok, sorry for insist. I have registered two ip phones using differents codecs (ulaw g729) and I have audio in both ways, so * is doing transcoding. But I am still getting the log 'Attempting native bridge of'.. so I wonder, What does that really mean? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attempting native bridge of
I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? The UAC and UAS are registering with * properly: --- sip.conf [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 - - dspam*CLI sip show peers Name/username HostDyn Nat ACL Port Status testgsm/testgsm172.16.51.244D 1 Unmonitored testg729/testg729 172.16.51.244D 2 Unmonitored -- Executing Answer(SIP/testgsm-081784b0, ) in new stack -- Executing Wait(SIP/testgsm-081784b0, 1) in new stack -- Executing Dial(SIP/testgsm-081784b0, SIP/testg729) in new stack -- Called testg729 -- SIP/testg729-0817dd90 is ringing -- SIP/testg729-0817dd90 answered SIP/testgsm-081784b0 -- Attempting native bridge of SIP/testgsm-081784b0 and SIP/testg729-0817dd90 - After the call is established the UAC is sending some RTP captured in a pcap file in gsm: -- tcpdump -T rtp udp --- 15:58:31.868404 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.868676 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.895551 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.895775 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936468 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936477 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936711 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936908 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 - Is there something wrong within the SDP? or Am I doing something wrong? Any comments would be appreciated.. thanks!! P.S. I am using Asterisk 1.2.12.1 if that matters. -- Greetings... Víctor Toofic --- 2006-11-15 16:15:12 UDP message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 138 v=0 o=user1 53655765 2353687637 IN IP4 172.16.51.244 s=- c=IN IP4 172.16.51.244 t=0 0 m=audio 10001 RTP/AVP 0 a=rtpmap:18 GSM/8000 --- 2006-11-15 16:15:12 UDP message received [404] bytes : SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received=172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- 2006-11-15 16:15:12 UDP message received [609] bytes : SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received=172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 161 v=0 o=root 3567 3567 IN IP4 172.16.51.215 s=session c=IN IP4 172.16.51.215 t=0 0 m=audio 17050 RTP/AVP 18 a=rtpmap:18 GSM/8000 a=silenceSupp:off - - - - --- 2006-11-15 16:15:12 UDP message sent: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-5 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 --- 2006-11-15 16:16:15 UDP message sent: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-9 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL
[asterisk-users] astertest
Hi all!! I've made some changes to the applications that Astertest was using to monitor the performance of the server. Now is also possible to track the bandwidth usage of the server, this has nothing to do with the executable (astertest.exe) itself but with the events that the Asterisk Manager generates. The method described in: http://www.asteriskguru.com/tutorials/astertest.html to perform the test is still valid. In the next days I am gonna make available some scripts to originate the calls and to make some graphs of the test, just like astertest does ;) You can find the sources here: http://toofic.no-ip.org/pub/src/app_securax.tar.gz I've compiled them against Asterisk 1.2.12.1, but I think there should not be problems with other versions. I hope someone could find it useful. -- Grettings, Víctor Toofic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax... Hi!! I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and :set filetype=asterisk :syntax on (optionally) works fine for me. -- Víctor Toofic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk+ser+docs
Where can I find docs on ser and asterisk intergration. Well, you could start googling around just a bit, but here you go anyway: I found these both useful: http://www.voip-info.org/wiki/view/OpenSER http://www.iptel.org/ser/doc/gettingstarted -- Victor Toofic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
