Re: [asterisk-users] MGCP
Hi! I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, You will most probably need 1.8 for this, with 1.4 you will certainly not be able to succed. Read more: http://www.voip-info.org/wiki/view/Asterisk+MGCP+channels Quote Matthew Fredrickson: A new channel driver, called chan_ccs, that allows, among other things, you to control MGCP media gateways for bearer trunks, instead of having to locally terminate them on the asterisk box that's controlling the signaling links. There is also code in the same branch that has chan_ccs that modified chan_mgcp so that Asterisk can act as a media gateway (since I don't have any good real media gateways to test on). This basically means you can have Asterisk TDM channel scalability up to (in the ideal state) the same level as you can do with SIP with no media, per box. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MGCP
Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MGCP
On Fri, Oct 29, 2010 at 4:21 AM, Baha @ SH i...@saudihome.com wrote: Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? In my research to try to get MEGACO protocol to work (they are very similar) I remember stumbling onto this information. I am not sure where or if it is even correct, it has been so long. MGCP is supported but only the MGCP phones, not the entire protocol. I tried to get full support for MEGCO because many NEC systems at that time used it for VoIP and the only other way to interface with an NEC IPK was via TDM, either POTS or T1, which are and especially were, WAY too expensive. I am out of date with NEC, I haven't touched one in years, but way back then, they were in the top three of PBX market share. You can probably google my name and MEGACO and you may find the info you are looking for. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MGCP
On Thu, Oct 28, 2010 at 9:54 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Oct 29, 2010 at 4:21 AM, Baha @ SH i...@saudihome.com wrote: Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? In my research to try to get MEGACO protocol to work (they are very similar) I remember stumbling onto this information. I am not sure where or if it is even correct, it has been so long. MGCP is supported but only the MGCP phones, not the entire protocol. I tried to get full support for MEGCO because many NEC systems at that time used it for VoIP and the only other way to interface with an NEC IPK was via TDM, either POTS or T1, which are and especially were, WAY too expensive. I am out of date with NEC, I haven't touched one in years, but way back then, they were in the top three of PBX market share. You can probably google my name and MEGACO and you may find the info you are looking for. Thanks, Steve Totaro Straight from the mouth of BKW three years ago. http://www.spinics.net/lists/asterisk/msg76756.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MGCP FXO endpoint
I have a fxo endpoint installed in a Cisco router. I would like in my dialplan to get an extension call a telephone number through that fxo endpoint. Since with zaptel channels it is done like: exten = 0999,1,Dial(DAHDI/2-1/111) -- being 111 the phone number I want to call. I thought that for mgcp it would be the same, and I did: exten = 5200,1,Dial(MGCP/aaln/S0/SU3/0...@armario11/111) aaln/S0/SU3/0 -- is an endpoint at ARMARIO11 The problem is that asterisk detects try to find host ARMARIO11/111 instead of calling number 111 in that FXO port. Here is the debug: -- Executing [5...@internal:1] Dial(MGCP/aaln/S0/SU2/0...@ignacio-1, MGCP/aaln/S0/SU3/0...@armario11/111) in new stack [Mar 5 13:36:50] NOTICE[4659]: chan_mgcp.c:1753 find_subchannel_and_lock: Gateway 'ARMARIO11/111' (and thus its endpoint 'aaln/S0/SU3/0') does not exist [Mar 5 13:36:50] WARNING[4659]: chan_mgcp.c:3541 mgcp_request: Unable to find MGCP endpoint 'aaln/S0/SU3/0...@armario11/111' [Mar 5 13:36:50] WARNING[4659]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'MGCP' (cause 20 - Unknown) Is there any way to achieve that? Thank you very much. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MGCP Thomson, early transmit problem
Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: user A - pstn --- ASTERISK - mgcp -- user B User A, connected behind a PSTN, tries to call User B. After dialing User B's number, call comes to ASTERISK, ASTERISK contacts User B. User B hears alerting/ringing tone, on display sees User A's number. User A hears ringback. But, ASTERISK(?) at the same time transmits sound received from User A to User B's phone, so User B hears User A!. Both of them are still hearing ringing/ringback tone. Call is not connected (from PSTN point of view). User B just additionaly hears sound from User A side. Do You have any idea, what's going on and if it's ASTERISK/Thomson ST2030/ feature or bug? In mgcp.conf I have something like this: [00147xxx] context=from-internal host=dynamic callerid=XXX 2992 threewaycalling=yes transfer=yes callwaiting=yes cancallforward=yes line = aaln/1 where 00147xxx is mac address of mgcp ip phone. Thanks Igor ___ -- 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] mgcp codec problem about ulaw
Hi: I have a mgcp.conf and a mgcp_additional.conf which records the special information about the extensions. And i found if i use ulaw in the general context in mgcp.conf,then all the registered extensions can make both outbound and inbound calls,the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw; can be disable and do no effect #include mgcp_additional.conf But if i disable ulaw and the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all ;allow=ulaw ; be disable and all the extensions can not be called allow=alaw; can be disable and do no effect #include mgcp_additional.conf then all the registered extensions can make outbound calls but can not be called by sip phone.The output of asterisk is following: asterisk1*CLI -- Executing NoOp(SIP/1000-084d94b8, here) in new stack -- Executing Dial(SIP/1000-084d94b8, MGCP/[EMAIL PROTECTED]|45) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/1000-084d94b8, ) in new stack == Spawn extension (from-internal, 6000, 3) exited non-zero on 'SIP/1000-084d94b8' -- Executing Macro(SIP/1000-084d94b8, hangupcall) in new stack -- Executing ResetCDR(SIP/1000-084d94b8, w) in new stack -- Executing NoCDR(SIP/1000-084d94b8, ) in new stack -- Executing GotoIf(SIP/1000-084d94b8, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf(SIP/1000-084d94b8, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait(SIP/1000-084d94b8, 5) in new stack -- Executing Hangup(SIP/1000-084d94b8, ) in new stack == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/1000-084d94b8' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/1000-084d94b8' the file mgcp_additional.conf is following: [6000] accountcode = 6000 context = [ext-local] callerid = 6000 6000 host = dynamic disallow = all allow = g723.1 allow = alaw allow = g729 allow = ulaw dtmfmode = rfc2833 nat = no line = 6000 the file extensions_additional.conf is following: [ext-local] exten = 6000,1,NoOp(here) exten = 6000,2,Dial(MGCP/[EMAIL PROTECTED],45) exten = 6000,3,Hangup exten = 6000,hint,MGCP/6000 Thanks in advance. ___ --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] mgcp registration with asterisk
HIi am trying to register mgcp gateways(Polycom 501, 601) to asterisk as a call agent, mgcp gateways are not registering to the call agent.Please help me on this if any one knows how to congigure the mgcp.conf on asterisk as well as an MGs.The following are the details of mgcp.conf on asterisk.mgcp.conf[general]port = 2427bindaddr = 0.0.0.0[0004f205c258] //MG MAC Addresshost = 172.21.67.137 //MG IP Address(static)context = defaultcanreinvite = noline = aaln/1Please Let me know if any one already tryied MGs registration with asterisk.Kind Regards,- Ashok P. Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --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] MGCP stuff - VoiPACK
On Wed, Oct 11, 2006 at 02:17:11PM -0400, Andrew Joakimsen wrote: Asterisk can only be the proxy/server for MGCP, you connect other devices to it. Asterisk can not be a user agent connecting to other MGCP server. As it happens, I have a 4 port VoiPACK gateway that speaks MGCP (legacy of an abortive FacetPhone trial. This thing seems insanely poorly documented; I gather fairly strongly that it is Chinese in manufacture and design, and OEM *only*. Anyone ever gotten one of them working with Asterisk? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] MGCP stuff
Hello everybody! I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol. What I want to do: I want to talk to the outside world via MGCP. I suppose I must set an MGCP peer to route outgoing calls. So, I must set the endpoint syntax of the Asterisk server (Asterisk will act as an MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP gateways via RTP. Ex: DALN/S1/SU0/0@my_address.mydomain.my_dns_suffix Where the part after @ is stored in BTS and contains my telephone number, etc (this is the providers problem). The question: is this possible with Asterisk? Where can I find some documentation for configuring mgcp.conf? The documentation (Asterisk: The Future of Telephony) says MGCP isnt completely developed. 10q! -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone:0744137020 email: [EMAIL PROTECTED] ___ --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] MGCP stuff
Asterisk can only be the proxy/server for MGCP, you connect other devices to it. Asterisk can not be a user agent connecting to other MGCP server. On 10/11/06, Paul Ianas [EMAIL PROTECTED] wrote: Hello everybody! I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol. What I want to do: I want to talk to the outside world via MGCP. I suppose I must set an MGCP peer to route outgoing calls. So, I must set the endpoint syntax of the Asterisk server (Asterisk will act as an MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP gateways via RTP. Ex: DALN/S1/SU0/0@my_address.mydomain.my_dns_suffix Where the part after @ is stored in BTS and contains my telephone number, etc (this is the provider's problem). The question: is this possible with Asterisk? Where can I find some documentation for configuring mgcp.conf? The documentation (Asterisk: The Future of Telephony) says MGCP isn't completely developed. 10q! -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] ___ --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 ___ --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] mgcp trouble
Download patch for ncs support in mgcp from http://asterisk.urtho.net/tiki-index.php. 2006/7/7, Christian Schnelle [EMAIL PROTECTED]: Hi,i try to use asterisk together with a webstar mgcp cable modem. if iconfigure the cable modem to act as a nuera v5.2 call agent it connectsto asterisk. but if i try to make a call the line will disabled.in order to get a deeper look here comes the logs:--snip--voip*CLIMGCP read:NTFY 694708904 aaln/[EMAIL PROTECTED] 192.168.100.207] MGCP 1.0 NCS 1.0N: [EMAIL PROTECTED]192.168.100.200]X: 619607CFO: hdfrom 192.168.100.207:2427 Verb: 'NTFY', Identifier: '694708904', Endpoint:'aaln/[EMAIL PROTECTED]192.168.100.207]', Version: 'MGCP 1.0'4 headers, 0 linesHandling request 'NTFY' on aaln/[EMAIL PROTECTED]Transmitting:200 694708904 OKto 192.168.100.207:2427-- Creating connection for aaln/[EMAIL PROTECTED] in cxmode:sendrecv callid: 3ac0d40c7e01a573We're at 192.168.100.200 port 32930Answering with capability 4Answering with capability 8Posting Request:CRCX 21 aaln/[EMAIL PROTECTED] 192.168.100.207] MGCP 1.0C: 3ac0d40c7e01a573L: p:20, a:PCMU, a:PCMAM: sendrecvX: 7e01a573v=0o=root 28921 28921 IN IP4 192.168.100.200 s=sessionc=IN IP4 192.168.100.200t=0 0m=audio 32930 RTP/AVP 0 8a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000to 192.168.100.207:2427 -- MGCP Asked to indicate tone: L/dl onaaln/[EMAIL PROTECTED]in cxmode: sendrecvPosting Request:RQNT 22 aaln/[EMAIL PROTECTED]192.168.100.207 ] MGCP 1.0X: 619607cfR: L/hu(N),L/hf(N),D/[0-9#*](N)S: L/dlto 192.168.100.207:2427-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED] ) created in state: DownMGCP read:200 21 OKI: 20D7v=0o=- 21 8407 IN IP4 192.168.100.207s=-c=IN IP4 192.168.100.207 t=0 0m=audio 53456 RTP/AVP 0 8a=X-pc-csuites-rtp:60/50a=X-pc-csuites-rtcp:80/70a=X-pc-nrekey:0a=sendrecva=mptime:20 20a=ptime:20from 192.168.100.207:2427 Verb: '200', Identifier: '21', Endpoint: 'OK', Version: '(null)'2 headers, 12 linesCapabilities: us - 12, them - 12, combined - 12Non-codec capabilities: us - 1, them - 0, combined - 0MGCP read: 518 22 Unsupported or unknown packagefrom 192.168.100.207:2427Verb: '518', Identifier: '22', Endpoint: 'Unsupported', Version: 'orunknown'1 headers, 0 lines --snap--the mgcp.conf is:--snip--;; MGCP Configuration for Asterisk;[general]port = 2727bindaddr = 0.0.0.0;prefcodec=G711Adisallow = all; The Codec list MUST be reduced by config or you will get an error 510 (protocol error)allow = ulaw; If this list includes Codec not listed in youATA configuration under CodecName:allow = alaw[ 192.168.100.207]context=defaulthost=192.168.100.207canreinvite=no;slowsequence = yesline = aaln/1line = aaln/2;line = *;[Softphone] ;context=default;host=192.168.100.46;line = aaln/1;;line = *[192.168.100.209]context=defaulthost= 192.168.100.209line = aaln/1line = aaln/2--snap--Could someone give me a hint?Thx in advance.--Best regardsChristian SchnelleHaving Trouble in Windows REBOOT!! Having Trouble in Linux BE ROOT !!___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Pagarbiai,Giedrius AugysSiauliu Universitetas, IST IP telefonijos inzinieriusTel. 8 41 590408Mob. Tel. 8 678 05790el. pastas [EMAIL PROTECTED] ___ --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] mgcp trouble
Hi, i try to use asterisk together with a webstar mgcp cable modem. if i configure the cable modem to act as a nuera v5.2 call agent it connects to asterisk. but if i try to make a call the line will disabled. in order to get a deeper look here comes the logs: --snip-- voip*CLI MGCP read: NTFY 694708904 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N: [EMAIL PROTECTED] X: 619607CF O: hd from 192.168.100.207:2427 Verb: 'NTFY', Identifier: '694708904', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 694708904 OK to 192.168.100.207:2427 -- Creating connection for aaln/[EMAIL PROTECTED] in cxmode: sendrecv callid: 3ac0d40c7e01a573 We're at 192.168.100.200 port 32930 Answering with capability 4 Answering with capability 8 Posting Request: CRCX 21 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 3ac0d40c7e01a573 L: p:20, a:PCMU, a:PCMA M: sendrecv X: 7e01a573 v=0 o=root 28921 28921 IN IP4 192.168.100.200 s=session c=IN IP4 192.168.100.200 t=0 0 m=audio 32930 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 to 192.168.100.207:2427 -- MGCP Asked to indicate tone: L/dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 22 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 619607cf R: L/hu(N),L/hf(N),D/[0-9#*](N) S: L/dl to 192.168.100.207:2427 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down MGCP read: 200 21 OK I: 20D7 v=0 o=- 21 8407 IN IP4 192.168.100.207 s=- c=IN IP4 192.168.100.207 t=0 0 m=audio 53456 RTP/AVP 0 8 a=X-pc-csuites-rtp:60/50 a=X-pc-csuites-rtcp:80/70 a=X-pc-nrekey:0 a=sendrecv a=mptime:20 20 a=ptime:20 from 192.168.100.207:2427 Verb: '200', Identifier: '21', Endpoint: 'OK', Version: '(null)' 2 headers, 12 lines Capabilities: us - 12, them - 12, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 MGCP read: 518 22 Unsupported or unknown package from 192.168.100.207:2427 Verb: '518', Identifier: '22', Endpoint: 'Unsupported', Version: 'or unknown' 1 headers, 0 lines --snap-- the mgcp.conf is: --snip-- ; ; MGCP Configuration for Asterisk ; [general] port = 2727 bindaddr = 0.0.0.0 ;prefcodec=G711A disallow = all ; The Codec list MUST be reduced by config or you will get an error 510 (protocol error) allow = ulaw; If this list includes Codec not listed in you ATA configuration under CodecName: allow = alaw [192.168.100.207] context=default host=192.168.100.207 canreinvite=no ;slowsequence = yes line = aaln/1 line = aaln/2 ;line = * ;[Softphone] ;context=default ;host=192.168.100.46 ;line = aaln/1 ;;line = * [192.168.100.209] context=default host=192.168.100.209 line = aaln/1 line = aaln/2 --snap-- Could someone give me a hint? Thx in advance. -- Best regards Christian Schnelle Having Trouble in Windows REBOOT!! Having Trouble in Linux BE ROOT !! ___ --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] MGCP Unable to find key
Although I do not have an answer I changed the title so maybe someone with MGCP experience may notice it. Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] -- Alejandro Vargas ___ --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 ___ --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] MGCP problem when through internet
I'm using mediatrix mgcp device without problems with [EMAIL PROTECTED] 2.0 over the LAN. But now I trying one of this devices through internet. My firs problem was nat, but I decided to leave this problem for later and try it through a vpn. I used gvpe because it is very transparent. The device connects ok but in many cases (I suspect when the bandwidth is low) the call is droped after a few seconds with a message like this: No command found on [192.168.0.105] for transaction 404. Ignoring.. The number (676 in this case) is variable. Isee with etereal that Asterisk is sending a message of RQNT 404 aaln/[EMAIL PROTECTED] MGCP 1.0 and the device responds 200 404 OK, but following this, the asterisk server starts rejecting the packets of the port it were using to communicate to the device. -- Alejandro Vargas ___ --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] MGCP service from Free Téléc om
I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free Télécom comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO interface. However, to improve quality, reliability, control, etc., I'd like to have Asterisk directly access the underlying MGCP service. Since this will take quite a bit of work (chan_mgcp presently acts only as Call Agent and cannot function as an endpoint), I first tried to configure an old Cisco ATA-186 to use the Free service. Although international and domestic long distance calls (both outgoing and incoming) work fine, there are problems with local calls. When calling some locations in Paris, the ATA user hears a severe echo (though there is no echo if Freebox is used). The 186, like most ATAs, has echo cancellation only for the analog line. That is working as expected; the remote party does not hear an echo. I would think that the far side echo would be canceled by the remote media gateway, but that does not seem to be the case. I don't believe that the caller has any control over this (the CA sends out requests and the endpoint obeys them), so it appears that the Freebox must be doing echo cancellation for both ends. Can someone confirm this? If it's true, is it possible for Asterisk to cancel echo from the remote end? On calls to nearby locations, such as my own POTS line or Free's voicemail service, there is no outgoing audio from the ATA. It appears to be a routing problem, because I can't ping these media gateways, typically 172.16.254.x, but can ping those where the audio is ok, typically 172.25.x.x. Packets do arrive *from* 172.16.254.x, and incoming audio is ok. However, the ATM protocol is RFC 1483 routed, VC mux, so there is no way to specify a gateway other than using the proper PVC, which I assume is 8/35 for all the private addresses used for VoIP, and 8/36 for Internet IPs. I'd like to see what the Freebox is doing differently, but don't know how, because this traffic does not appear on its Ethernet port. Is there a reasonably inexpensive tool that can monitor the packets on a DSL line? Or some other way to find out what is happening? Thanks in advance, Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP service from Free Téléc om
On Sat, 17 Sep 2005, Stewart Nelson wrote: I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free Télécom comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO interface. However, to improve quality, reliability, control, etc., I'd like to have Asterisk directly access the underlying MGCP service. Somewhat off topic, who makes the hardware that Free is using? Is it available/being used in the US? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP service from Free Télécom
Somewhat off topic, who makes the hardware that Free is using? Is it available/being used in the US? http://www.broadcom.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mgcp fon behind NAT gw
Hi I've a mgcp fon (swissvoice IP10s) behind a NAT router. Configured is NAT for both in/out going on port 2427. Now I got the following mgcp debug messages when i try mgcp audit endpoint endpoint -- from 172.16.98.57:2427 Verb: 'RSIP', Identifier: '5346', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Retransmitting: 200 5346 OK to 192.168.2.3:2427 MGCP read: RSIP 5346 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart - IMHO means 200 everything ok. But what means the RM: restart ? For me it look's like the asterisk knows the NAT gw and also the EP. But the EP can't find the Call Agent. It's clear at the display. waiting for call agent... the mgcp.conf looks like this -- [192.168.2.3] ;router als RGW context=default host=192.168.2.3 nat=yes line = aaln/1 callerid=1423 port 2427 is bound. Directly connected via switch there's no problem. May be anyone have some hints or tipps... thx in advance regards mathias roehl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp fon behind NAT gw
Am Mo, den 04.07.2005 schrieb Mathias Röhl um 13:40: ok, *done*, my fault, error in NAT configuration... regards mathias roehl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Groups
I am looking into using a Cisco T1 device that uses MGCP. Asterisk is talking to it fine, but I am having a hard time figuring out how to handle channel grouping like Zap does. With Zap, I can take channels 1-23 and make a group g1 out of it and then simply dial Zap/g1. Does MGCP have this type of functionality? Everything I've tried points to no... Thanks! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and SIP clients
Hi folks I seted up the asterisk with an active ISDN B1 AVM Card (german vendor) and it works fine, various SIP clients (IP fon snom, xlite under MacOSX) and also incoming and outgoing connectins. Ok. No problem. After that I configured a CP7940G with a MGCP IOS. It's connected to the asterisk too via switch. No NAT deivce between. And of course, I can call out to PSTN and also to one of the SIP clients. No problem. But when I try to call the MGCP fon from a SIP client or from outside, I get this error message. --snip *CLI Jun 20 15:30:20 NOTICE[25423]: chan_mgcp.c:1509 find_subchannel: Gateway '172.16.103.20' (and thus its endpoint 'd001/1') does not exist Jun 20 15:30:20 WARNING[25423]: chan_mgcp.c:3368 mgcp_request: Unable to find MGCP endpoint 'd001/[EMAIL PROTECTED]' Jun 20 15:30:20 NOTICE[25423]: app_dial.c:759 dial_exec: Unable to create channel of type 'MGCP' snip- Otherwise, vice versa it looks good. snip- Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.98.52:5060;branch=z9hG4bK0fe6a009 From: 1412 sip:[EMAIL PROTECTED];tag=as2ae905a1 To: sip:[EMAIL PROTECTED]:5060;line=xo91ds1x;tag=ycfd6114eu Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=xo91ds1x Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 -snip But I'm wondering why, the number 1412 is the MGCP fon. But it works. And as an additional info, why must I use with the Cisco CP7940 channel d001 and not (like with an IP10s swissvoice...) aaln ? may be someone can explain me. If there are missing infos please let me know best regards thx in advance mathias ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP files for Polycom
Title: MGCP files for Polycom Does anybody know were I can download the MGCP files for the Polycom IP500? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Useragent
Hi, -Original Message- 1- Anybody implement mgcp useragent in *. Nope. Hasn't been done yet. 2- Where can i get that. Not available in your nearest drugstore. 3- if no then anybody can help me to write it down. Digium ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Useragent
Hi 1- Anybody implement mgcp useragent in *. 2- Where can i get that. 3- if no then anybody can help me to write it down. Best Regards Ibrar Ahmed Project Manager. Comcept (Pvt) Ltd. Islamabad Pakistan www.com-cept.com [EMAIL PROTECTED] [EMAIL PROTECTED] Ph # (Off) +92-51-111784784 Ph # (Res) +92-51-2271283 Ph # (Mob) +92-3009543001 Fax # 92-51-111784785 www.com-cept.com Pick battles that are big enough to matter, small enough to win __ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and missing digit map
Hi all, I'm trying to setup an MGCP connection between my asterisk and a third party pbx system. I have very little control over the external pbx. The calls are failing with the following asterisk error: notice chan_mgcp.c 2347 handle_repsonse: Terminating on result 519 from aaln/[EMAIL PROTECTED] Searching the net shows that the result 519 relates to a digit map. It seems the pbx is requesting a digit map from Asterisk and failing because it is not getting one. Does asterisk/mgcp support this digit map feature? If it does what are the config settings? If asterisk does not support this feature what changes should I be asking the pbx support person to make to his system? I'm running fedora core 1 and Asterisk CVS-D2004.12.21.13.00 thanks in advance bruce ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP 1.0 / NCS 1.0
I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp show endpoints Gateway '001099008521' at 10.1.22.81 (Static) -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle *CLI mgcp audit debug no reload show *CLI mgcp audit No such command 'mgcp audit' (type 'help' for help) *CLI mgcp audit endpoint Usage: mgcp audit endpoint endpointid Lists the capabilities of an endpoint in the MGCP (Media Gateway Control Protocol) subsystem. mgcp debug MUST be on to see the results of this command. *CLI mgcp audit endpoint 1 Could not find endpoint *CLI mgcp audit endpoint aaln/1 Could not find endpoint *CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 9 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 9 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '9', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: RSIP 1075 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 338 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1075', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1075 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 10 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 3faf4f29 R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 11 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 10 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '10', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 11 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '11', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 11. Ignoring... MGCP read: RSIP 1076 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 354 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1076', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1076 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 12 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 696be0ea R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 13 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 12 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '12', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 13 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '13', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 13. Ignoring... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP 1.0 / NCS 1.0
Right, your ATA (cable modem?) is using the NCS 1.0 profile of MGCP. Asterisk does not currently support the NCS profile from cable labs. It would be very nice to see someday, as well as backwards support for MGCP 0.1 Duane Cox - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, May 20, 2005 2:52 PM Subject: [Asterisk-Users] MGCP 1.0 / NCS 1.0 I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp show endpoints Gateway '001099008521' at 10.1.22.81 (Static) -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle *CLI mgcp audit debug no reload show *CLI mgcp audit No such command 'mgcp audit' (type 'help' for help) *CLI mgcp audit endpoint Usage: mgcp audit endpoint endpointid Lists the capabilities of an endpoint in the MGCP (Media Gateway Control Protocol) subsystem. mgcp debug MUST be on to see the results of this command. *CLI mgcp audit endpoint 1 Could not find endpoint *CLI mgcp audit endpoint aaln/1 Could not find endpoint *CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 9 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 9 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '9', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: RSIP 1075 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 338 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1075', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1075 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 10 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 3faf4f29 R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 11 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 10 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '10', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 11 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '11', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 11. Ignoring... MGCP read: RSIP 1076 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 354 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1076', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1076 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 12 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 696be0ea R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 13 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 12 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '12', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 13 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '13', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 13. Ignoring... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP 1.0 / NCS 1.0
Check this here: http://asterisk.urtho.net/tiki-index.php On 5/20/05, Duane Cox [EMAIL PROTECTED] wrote: Right, your ATA (cable modem?) is using the NCS 1.0 profile of MGCP. Asterisk does not currently support the NCS profile from cable labs. It would be very nice to see someday, as well as backwards support for MGCP 0.1 Duane Cox - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, May 20, 2005 2:52 PM Subject: [Asterisk-Users] MGCP 1.0 / NCS 1.0 I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp show endpoints Gateway '001099008521' at 10.1.22.81 (Static) -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle *CLI mgcp audit debug no reload show *CLI mgcp audit No such command 'mgcp audit' (type 'help' for help) *CLI mgcp audit endpoint Usage: mgcp audit endpoint endpointid Lists the capabilities of an endpoint in the MGCP (Media Gateway Control Protocol) subsystem. mgcp debug MUST be on to see the results of this command. *CLI mgcp audit endpoint 1 Could not find endpoint *CLI mgcp audit endpoint aaln/1 Could not find endpoint *CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 9 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 9 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '9', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: RSIP 1075 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 338 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1075', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1075 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 10 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 3faf4f29 R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 11 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 10 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '10', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 11 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '11', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 11. Ignoring... MGCP read: RSIP 1076 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 354 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1076', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1076 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 12 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 696be0ea R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 13 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 12 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '12', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 13 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '13', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 13. Ignoring... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.aefirion.org/ http
RE: [Asterisk-Users] MGCP 1.0 / NCS 1.0
Unfortunately I don't think my ATA falls into any of these... We have an ATA made by Innomedia MTA-3328R -- it acts like a standard ATA but with MGCP / NCS 1.0... I was thinking that if I could sniff the connection to the gatekeeper - I could write a channel to support this? _ From: [EMAIL PROTECTED] on behalf of Michael Bielicki Sent: Fri 5/20/2005 7:34 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] MGCP 1.0 / NCS 1.0 Check this here: http://asterisk.urtho.net/tiki-index.php http://asterisk.urtho.net/tiki-index.php On 5/20/05, Duane Cox [EMAIL PROTECTED] wrote: Right, your ATA (cable modem?) is using the NCS 1.0 profile of MGCP. Asterisk does not currently support the NCS profile from cable labs. It would be very nice to see someday, as well as backwards support for MGCP 0.1 Duane Cox - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, May 20, 2005 2:52 PM Subject: [Asterisk-Users] MGCP 1.0 / NCS 1.0 I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp show endpoints Gateway '001099008521' at 10.1.22.81 (Static) -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle -- 'aaln/[EMAIL PROTECTED] in 'default' is idle *CLI mgcp audit debug no reload show *CLI mgcp audit No such command 'mgcp audit' (type 'help' for help) *CLI mgcp audit endpoint Usage: mgcp audit endpoint endpointid Lists the capabilities of an endpoint in the MGCP (Media Gateway Control Protocol) subsystem. mgcp debug MUST be on to see the results of this command. *CLI mgcp audit endpoint 1 Could not find endpoint *CLI mgcp audit endpoint aaln/1 Could not find endpoint *CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 9 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 9 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '9', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: RSIP 1075 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 338 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1075', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1075 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 10 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 3faf4f29 R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 11 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 10 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '10', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 11 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '11', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521] for transaction 11. Ignoring... MGCP read: RSIP 1076 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: disconnected RD: 354 from 10.1.22.81:2427 Verb: 'RSIP', Identifier: '1076', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'RSIP' on aaln/[EMAIL PROTECTED] -- Resetting interface aaln/[EMAIL PROTECTED] Transmitting: 200 1076 OK to 10.1.22.81:2427 -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 12 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 696be0ea R: L/hd(N) to 10.1.22.81:2427 Posting Request: AUEP 13 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.1.22.81:2427 MGCP read: 528 12 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '12', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines MGCP read: 528 13 Incompatible Protocol Version from 10.1.22.81:2427 Verb: '528', Identifier: '13', Endpoint: 'Incompatible', Version: 'Protocol Version' 1 headers, 0 lines -- No command found on [001099008521
RE: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
I solved the problem by rechecking my configuration files, namely mgcp.conf and extensions.conf. I changed the EPIDx strings in the ATA188 to a001 and a002 (and changed accordingly in other config files), the context from default to ext_mgcp in mgcp.conf and set all the ports to 2427 and now it works. Hope this helps. JFD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 10 mai 2005 12:07 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel Nevermind, I have solved the problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 10 mai 2005 10:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel Importance: High When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to * (IP 192.168.1.59, port 2727). On the other hand, after sending 2 packets at startup, * does not respond to the ATA188. This certainly looks like a configuration problem, but I just can't seem to find exactly what is wrong. If someone has experienced the same problem or know what is wrong then I would really appreciate your help. Thanks, JF Modules.conf : [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so ;noload = res_musiconhold.so noload = app_festival.so noload = app_url.so noload = app_image.so noload = app_disa.so noload = app_qcall.so noload = app_adsiprog.so noload = app_ices.so ;noload = codec_g726.so ;noload = codec_alaw.so ;noload = format_vox.so ;noload = format_h263.so noload = format_jpeg.so ;noload = cdr_csv.so ;noload = cdr_manager.so ;noload = app_zapras.so ;noload = app_flash.so ;noload = app_zapbarge.so ;noload = app_zapscan.so ;noload = app_talkdetect.so ;noload = app_alarmreceiver.so ;noload = chan_skinny.so ;noload = chan_sip.so noload = chan_alsa.so ;noload = chan_oss.so [global] chan_modem.so=yes Mgcp.conf : [general] port = 2727 bindaddr = 192.168.1.59 [MGCP1] context=default host=192.168.1.27 line=aaln/1 line=aaln/2 [MGCP2] context=default host=192.168.1.28 line=aaln/1 line=aaln/2 Extensions.conf : [general] static = yes writeprotect = no [globals] TMGCP1=MGCP/aaln/[EMAIL PROTECTED] TMGCP2=MGCP/aaln/[EMAIL PROTECTED] TMGCP3=MGCP/aaln/[EMAIL PROTECTED] TMGCP4=MGCP/aaln/[EMAIL PROTECTED] TSIP1=SIP/SIP1 TSIP2=SIP/SIP2 [default] exten = 70,1,Dial(${TMGCP1},20,tr) exten = 71,1,Dial(${TMGCP2},20,tr) exten = 72,1,Dial(${TMGCP3},20,tr) exten = 73,1,Dial(${TMGCP4},20,tr) exten = 74,1,Dial(${TSIP1},20,tr) exten = 75,1,Dial(${TSIP2},20,tr) ATA188 config : Cisco ATA 188 (MGCP) Configuration : UIPassword: * UseTftp:1 TftpURL:0 CfgInterval: 3600 EncryptKey: * EncryptKeyEx: Dhcp: 0 StaticIP: 192.168.1.27 StaticRoute: 192.168.1.1 StaticNetMask: 255.255.255.0 EPID0orSID0: . EPID1orSID1: . CA0orCM0: 192.168.1.59:2727 CA1orCM1: 0 CA0UID: 0 CA1UID: 0 MGCPVer: NCS1.0 RetxIntvl: 500 RetxLim: 10 MGCPPort: 2427 CodecName: PCMU,PCMA,G723,G729 LBRCodec: 3 PrfCodec: 1 AudioMode: 0x00350035 ConnectMode: 0x9400 CallerIdMethod: 0xc0019e60 DNS1IP: 0.0.0.0 DNS2IP: 0.0.0.0 Domain: . NumTxFrames: 2 TOS: 0x68b8 OpFlags: 0x0002 VLANSetting: 0x002b Polarity: 0x FXSInputLevel: 0 FXSOutputLevel: -4 SigTimer: 0x0064 RingCadence: 2,4,25 DialTone: 2,31538,30831,1380,1740,1,0,0,1000,0,0 BusyTone: 2,30467,28959,1191,1513,0,4000,4000,0,0,0 ReorderTone: 2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0 RingBackTone: 2,30831,30467,1943,2111,0,16000,32000,0,0,0 CallWaitTone: 1,30831,0,5493,0,0,2400,2400,4800,0,0 AlertTone: 1,30467,0,5970,0,0,480,480,1920,0,0 NPrintf: 0.0.0.0.0 TraceFlags: 0x SyslogIP: 0.0.0.0.514 SyslogCtrl: 0x MediaPort: 16384 CFGID: 0x ata0013199e70f5 Version: v3.1.1 atamgcp (Build 040629A) Features: 0x0017 HardwareVersion: 0x0010 0x ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users
RE: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
Nevermind, I have solved the problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 10 mai 2005 10:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel Importance: High When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to * (IP 192.168.1.59, port 2727). On the other hand, after sending 2 packets at startup, * does not respond to the ATA188. This certainly looks like a configuration problem, but I just can't seem to find exactly what is wrong. If someone has experienced the same problem or know what is wrong then I would really appreciate your help. Thanks, JF Modules.conf : [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so ;noload = res_musiconhold.so noload = app_festival.so noload = app_url.so noload = app_image.so noload = app_disa.so noload = app_qcall.so noload = app_adsiprog.so noload = app_ices.so ;noload = codec_g726.so ;noload = codec_alaw.so ;noload = format_vox.so ;noload = format_h263.so noload = format_jpeg.so ;noload = cdr_csv.so ;noload = cdr_manager.so ;noload = app_zapras.so ;noload = app_flash.so ;noload = app_zapbarge.so ;noload = app_zapscan.so ;noload = app_talkdetect.so ;noload = app_alarmreceiver.so ;noload = chan_skinny.so ;noload = chan_sip.so noload = chan_alsa.so ;noload = chan_oss.so [global] chan_modem.so=yes Mgcp.conf : [general] port = 2727 bindaddr = 192.168.1.59 [MGCP1] context=default host=192.168.1.27 line=aaln/1 line=aaln/2 [MGCP2] context=default host=192.168.1.28 line=aaln/1 line=aaln/2 Extensions.conf : [general] static = yes writeprotect = no [globals] TMGCP1=MGCP/aaln/[EMAIL PROTECTED] TMGCP2=MGCP/aaln/[EMAIL PROTECTED] TMGCP3=MGCP/aaln/[EMAIL PROTECTED] TMGCP4=MGCP/aaln/[EMAIL PROTECTED] TSIP1=SIP/SIP1 TSIP2=SIP/SIP2 [default] exten = 70,1,Dial(${TMGCP1},20,tr) exten = 71,1,Dial(${TMGCP2},20,tr) exten = 72,1,Dial(${TMGCP3},20,tr) exten = 73,1,Dial(${TMGCP4},20,tr) exten = 74,1,Dial(${TSIP1},20,tr) exten = 75,1,Dial(${TSIP2},20,tr) ATA188 config : Cisco ATA 188 (MGCP) Configuration : UIPassword: * UseTftp:1 TftpURL:0 CfgInterval: 3600 EncryptKey: * EncryptKeyEx: Dhcp: 0 StaticIP: 192.168.1.27 StaticRoute: 192.168.1.1 StaticNetMask: 255.255.255.0 EPID0orSID0: . EPID1orSID1: . CA0orCM0: 192.168.1.59:2727 CA1orCM1: 0 CA0UID: 0 CA1UID: 0 MGCPVer: NCS1.0 RetxIntvl: 500 RetxLim: 10 MGCPPort: 2427 CodecName: PCMU,PCMA,G723,G729 LBRCodec: 3 PrfCodec: 1 AudioMode: 0x00350035 ConnectMode: 0x9400 CallerIdMethod: 0xc0019e60 DNS1IP: 0.0.0.0 DNS2IP: 0.0.0.0 Domain: . NumTxFrames: 2 TOS: 0x68b8 OpFlags: 0x0002 VLANSetting: 0x002b Polarity: 0x FXSInputLevel: 0 FXSOutputLevel: -4 SigTimer: 0x0064 RingCadence: 2,4,25 DialTone: 2,31538,30831,1380,1740,1,0,0,1000,0,0 BusyTone: 2,30467,28959,1191,1513,0,4000,4000,0,0,0 ReorderTone: 2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0 RingBackTone: 2,30831,30467,1943,2111,0,16000,32000,0,0,0 CallWaitTone: 1,30831,0,5493,0,0,2400,2400,4800,0,0 AlertTone: 1,30467,0,5970,0,0,480,480,1920,0,0 NPrintf: 0.0.0.0.0 TraceFlags: 0x SyslogIP: 0.0.0.0.514 SyslogCtrl: 0x MediaPort: 16384 CFGID: 0x ata0013199e70f5 Version: v3.1.1 atamgcp (Build 040629A) Features: 0x0017 HardwareVersion: 0x0010 0x ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP : chan_mgcp.c:1509 find_subchannel
[EMAIL PROTECTED] schrieb: Nevermind, I have solved the problem. It would be nice to know how! I have the same message, and i want to get rid of it, too. Best regards Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP issue
you need make change in indications.conf wells On 3/25/05, Daniel Eboa [EMAIL PROTECTED] wrote: Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = test 8000100 callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line = aaln/1 callerid= test 8000101 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/2 callerid= test 8000102 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/3 callerid= test 8000104 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/4 extensions.conf [MGCP] include = Toll Free include = CreoLink exten = 8000100,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000100,2,Hangup exten = 8000101,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000101,2,Hangup exten = 8000102,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000102,2,Hangup exten = 8000103,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000103,2,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP issue
What exactly should I need to change in indications.conf?? Thanks. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of wells zheng Sent: Wednesday, May 04, 2005 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MGCP issue you need make change in indications.conf wells On 3/25/05, Daniel Eboa [EMAIL PROTECTED] wrote: Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = test 8000100 callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line = aaln/1 callerid= test 8000101 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/2 callerid= test 8000102 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/3 callerid= test 8000104 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/4 extensions.conf [MGCP] include = Toll Free include = CreoLink exten = 8000100,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000100,2,Hangup exten = 8000101,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000101,2,Hangup exten = 8000102,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000102,2,Hangup exten = 8000103,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000103,2,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and CISCO 7960?
Is someone running mgcp firmware with asterisk? I need to verify the phone issues Thanks. Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP issue
Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = test 8000100 callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line = aaln/1 callerid= test 8000101 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/2 callerid= test 8000102 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/3 callerid= test 8000104 callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line = aaln/4 extensions.conf [MGCP] include = Toll Free include = CreoLink exten = 8000100,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000100,2,Hangup exten = 8000101,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000101,2,Hangup exten = 8000102,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000102,2,Hangup exten = 8000103,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten = 8000103,2,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Channel Lockup and other probelms
post your mgcp.conf (mask out any ips if you need to) duane cox - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 16, 2005 3:00 PM Subject: [Asterisk-Users] MGCP Channel Lockup and other probelms Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully completes a call. 2. While trying to get the dialing to work. The MGCP channel hung. the call came in IAX.. I hung up the originating channel (IAX side) and it never reported the hangup.. now I see: Mar 16 14:12:12 WARNING[8904]: channel.c:523 ast_channel_walk_locked: Avoided deadlock for 'MGCP/aaln/[EMAIL PROTECTED]', 10 retries! over and over.. Show channels shows: IAX2/[EMAIL PROTECTED]/2 (cp-austin 5125551234 1 )Ring Dial MGCP/aaln/[EMAIL PROTECTED] soft hangup on the IAX side dosen't do anything soft hangup on the MGCP side things it's not a valid channel tab completion in the CLI seems to be sluggish and locks the CLI sometimes (have to ctrl-c and go back in with asterisk -r) mgcp reload doesn't do anything (now it says it's waiting for the previous reload to finish) unload chan_mgcp.so says it's being used unload -f chan_mgcp.so says it failed. haven't tried unload -h yet.. scared.. Any ideas? what may have caused these problems and how do I get out of this mess without restarting the whole system? BTW, SIP calls are completing just fine.. Thanks, Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Channel Lockup and other probelms
Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully completes a call. 2. While trying to get the dialing to work. The MGCP channel hung. the call came in IAX.. I hung up the originating channel (IAX side) and it never reported the hangup.. now I see: Mar 16 14:12:12 WARNING[8904]: channel.c:523 ast_channel_walk_locked: Avoided deadlock for 'MGCP/aaln/[EMAIL PROTECTED]', 10 retries! over and over.. Show channels shows: IAX2/[EMAIL PROTECTED]/2 (cp-austin 5125551234 1 )Ring Dial MGCP/aaln/[EMAIL PROTECTED] soft hangup on the IAX side dosen't do anything soft hangup on the MGCP side things it's not a valid channel tab completion in the CLI seems to be sluggish and locks the CLI sometimes (have to ctrl-c and go back in with asterisk -r) mgcp reload doesn't do anything (now it says it's waiting for the previous reload to finish) unload chan_mgcp.so says it's being used unload -f chan_mgcp.so says it failed. haven't tried unload -h yet.. scared.. Any ideas? what may have caused these problems and how do I get out of this mess without restarting the whole system? BTW, SIP calls are completing just fine.. Thanks, Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP howto
Hey there, I'm an asterisk newbie and have just joined this mailing list. I have to use asterisk as a call agent that supports MGCP requests. I'm reading the documentation from asteriskdocs and voip-info.org but those cover more specifically only IAX and SIP configuration. I'd really appreciate it if someone can tell me where to find more detailed documentation on how to configure asterisk to work with MGCP. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP howto
Hey there, I'm an asterisk newbie and have just joined this mailing list. I have to use asterisk as a call agent that supports MGCP requests. I'm reading the documentation from asteriskdocs and voip-info.org but those cover more specifically only IAX and SIP configuration. I'd really appreciate it if someone can tell me where to find more detailed documentation on how to configure asterisk to work with MGCP. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP howto
On Mon, 7 Mar 2005, Fabio Margarido wrote: I'm an asterisk newbie and have just joined this mailing list. I have to use asterisk as a call agent that supports MGCP requests. I'm reading the documentation from asteriskdocs and voip-info.org but those cover more specifically only IAX and SIP configuration. I'd really appreciate it if someone can tell me where to find more detailed documentation on how to configure asterisk to work with MGCP. You really are going to have to expand on what you need to know which can't be gleaned from reading mgcp.conf - there is no other documentation apart from that and past discussion on here. BTW just to get it out of the way, Asterisk does not have a MGCP User Agent - *only* Call Agent. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP to Inter Tel system
I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... Dustin Moore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP to Inter Tel system
Dustin Moore wrote: I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... The short answer: NO. Asterisk does not act as an MGCP endpoint, only as a call-agent. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP to Inter Tel system
On Thu, 2005-03-03 at 12:23 -0800, Dustin Moore wrote: I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... The archives are searchable. Use google and learn about the advanced options such as limiting to a site. Leo answered the other part, MGCP is modeled much like normal phone networks where there are 2 distinct roles and they talk only to the other role. Asterisk and the InterTel play the same role and therefore can't talk to each other. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP transfer and CDR
Hello: I have an asterisk deployment with 15 MGCP extensions and 30 incoming E1 R2 channels. Calls are received by a receptionist queue (which only member is the receptionist phone). The receptionist then transfers (using hook flash) the call to one of the extensions. I want to be able to have two separate records in the cdr, one for the call between the receptionist and the unicall channel and another that starts when the call is transferred. I have tried to use forkcdr in the extensions context, seems to work randomly. Also call records show that the destination is between the receptionist and the extension or between the unicall channel and the extension. Sometimes minutes are billed to the receptionist, sometimes to the call between extension. I want cdr to show dst and src correctly. Any Ideas? thanks, ia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP - Unicall
Hi: I have installed a call center with 15 extensions. I used an E1 with R2 signalling for incoming calls. The calls are received through clarent cpg-101 (aka DLINK 104s). I have a lot of trouble with transfers. Usually an extension receives the incoming calls from the e1 then the call is transferred (using flash key) to another extension. Behavior of transfers seem to be erratic sometimes the transfer goes OK. But sometimes when call is transferred the person in the extension hears the caller from the E1 but the caller does not hears the person in the extension. I suspect that the problem is the mgcp part (either asterisk or the device) because the e1 calls go ok before they are transferred. I search the list and found posts about firmware upgrades. But i cannot find ther firmware upgrade and i am afraid that since i am using the original clarent device flashing with the dlink firmware may render it useless. any ideas? thanks, ia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP, Asterisk Cisco VG200
Hello: I want to receive calls from my SIP proxy and re-route them to one of the analog lines on my Cisco VG200 ia MGCP and Asterisk. Inbound SIP calls will arrive with the five digit called number preficed with an m by the proxy. I'd like to them match these calls against a rule like exten = _mX,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) . This however results in an error. My mgcp.conf looks like: ; mgcp audit endpoint aaln/[EMAIL PROTECTED] (vg200) [general] port = 2427 bindaddr = 128.100.10.10 (asterisk server) [vg200] host = 128.100.10.11 canreinvite = no line = aaln/2 line = aaln/1 Can anyone explain how this should work? Thanks,Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP, Asterisk Cisco VG200
See my comments inline: snip ; mgcp audit endpoint aaln/[EMAIL PROTECTED] (vg200) [general] port = 2427 bindaddr = 128.100.10.10 (asterisk server) [vg200] Whatever host name that you put here should be resolvable either by /etc/hosts or DNS lookup. If not, set it to the IP address of the host, i.e. [128.100.10.11]. Hope this helps. snip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP phone
Does anyone know somefreeMGCP softphone? Nowadays I'm using one from eyeP Media, but it is trial for 30 days and it's expiring... Any ideas? Thanks, Leonardo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP parameters
On Fri, Dec 31, 2004 at 10:03:54AM -0300, Leonardo J. Tramontina wrote: 1) I am using Asterisk and a softphone called MGCP eyeP Phone. I was watching the traffic between them at Ethereal and observed that some of them have extra parameters. Example: CreateConnection has Request Identifier (X), that is not described on RFC 2705. Should I ignore or consider it? I believe X is spec'd in RFC 3435. 2) There are some parameters that don't have identifier? I mean, Request Identifier is X:; Observed Events is O: ; Call ID is C: ... These are them: - Notified Entity - Remote Connection Descriptor - encapsulated Endpoint Configuration - Second Endpoint Id - encapsulated Notification Request See above. 2.1) How can I identify them when they exist? See above. RFCs 2705 and 3435 both specify how to handle unknown commands and parameters. 2.2) What means encapsulated parameters?? When you receive a RQNT (request for notify) you include the action to be taken for different types of events. The action can be a stripped down RQNT that provides new instructions, just like a new RQNT packet. This can include parameters. I've heard it call encapsulated parameters but I believe that the spec calls it embedded. YMMV. FWIW, Asterisk's MGCP support is a bit dubious and definitely not what the spec intended. The spec intended for a call agent to manage various media gateways and orchestrate what was going on. Asterisk is a media gateway that attempts to be a call agent for itself and another media gateway (the phone) with interesting but limited results. It probably will never operate in a real MGCP environment unless chan_mgcp is implemented as a media gateway. Right now, it just provides a hack that allows MGCP phones to hit it...sometimes. Good luck. -- Jayson Vantuyl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP parameters
Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters -- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) ReturnCode, -- DeleteConnection( CallId, EndpointId, ConnectionId, Reason-code, Connection-parameters) ReturnCode, -- DeleteConnection( CallId, EndpointId) ReturnCode, -- DeleteConnection( EndpointId) ReturnCode, EndPointIdList|{ [RequestedEvents,] [DigitMap,] [SignalRequests,] [RequestIdentifier,] [NotifiedEntity,] [ConnectionIdentifiers,] [DetectEvents,] [ObservedEvents,] [EventStates,] [BearerInformation,] [RestartReason,] [RestartDelay,] [ReasonCode,] [Capabilities]} --- AuditEndPoint(EndpointId, [RequestedInfo]) ReturnCode, [CallId,] [NotifiedEntity,] [LocalConnectionOptions,] [Mode,] [RemoteConnectionDescriptor,] [LocalConnectionDescriptor,] [ConnectionParameters] --- AuditConnection(EndpointId, ConnectionId, RequestedInfo) ReturnCode, [NotifiedEntity] --- RestartInProgress ( EndPointId, RestartMethod, [RestartDelay,] [Reason-code]) ReturnCode -- EndpointConfiguration( EndpointId, BearerInformation) ReturnCode -- NotificationRequest( EndpointId, [NotifiedEntity,] [RequestedEvents,] RequestIdentifier, [DigitMap,] [SignalRequests,] [QuarantineHandling,] [DetectEvents,] [encapsulated EndpointConfiguration]) ReturnCode -- Notify( EndpointId, [NotifiedEntity,] RequestIdentifier, ObservedEvents) ReturnCode, ConnectionId, [SpecificEndPointId,] [LocalConnectionDescriptor,] [SecondEndPointId,] [SecondConnectionId] --- CreateConnection(CallId, EndpointId, [NotifiedEntity,] [LocalConnectionOptions,] Mode, [{RemoteConnectionDescriptor | SecondEndpointId}, ] [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) ReturnCode, [LocalConnectionDescriptor] --- ModifyConnection(CallId, EndpointId, ConnectionId, [NotifiedEntity,] [LocalConnectionOptions,] [Mode,] [RemoteConnectionDescriptor,] [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) My questions: 1) I am using Asterisk and a softphone called MGCP eyeP Phone. I was watching the traffic between them at Ethereal and observed that some of them have extra parameters. Example: CreateConnection has Request Identifier (X), that is not described on RFC 2705. Should I ignore or consider it? 2) There are some parameters that don't have identifier? I mean, Request Identifier is X:; Observed Events is O: ; Call ID is C: ... These are them: - Notified Entity - Remote Connection Descriptor - encapsulated Endpoint Configuration - Second Endpoint Id - encapsulated Notification Request 2.1) How can I identify them when they exist? 2.2) What means encapsulated parameters?? Thanks in advance, Leonardo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] MGCP parameters
The RFC specification alone is not sufficient, there are many signaling packages that are defined elsewhere. Also, RFC 2705 is out of date, see RFC 3435 Leonardo J. Tramontina wrote: Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters -- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) ReturnCode, -- DeleteConnection( CallId, EndpointId, ConnectionId, Reason-code, Connection-parameters) ReturnCode, -- DeleteConnection( CallId, EndpointId) ReturnCode, -- DeleteConnection( EndpointId) ReturnCode, EndPointIdList|{ [RequestedEvents,] [DigitMap,] [SignalRequests,] [RequestIdentifier,] [NotifiedEntity,] [ConnectionIdentifiers,] [DetectEvents,] [ObservedEvents,] [EventStates,] [BearerInformation,] [RestartReason,] [RestartDelay,] [ReasonCode,] [Capabilities]} --- AuditEndPoint(EndpointId, [RequestedInfo]) ReturnCode, [CallId,] [NotifiedEntity,] [LocalConnectionOptions,] [Mode,] [RemoteConnectionDescriptor,] [LocalConnectionDescriptor,] [ConnectionParameters] --- AuditConnection(EndpointId, ConnectionId, RequestedInfo) ReturnCode, [NotifiedEntity] --- RestartInProgress ( EndPointId, RestartMethod, [RestartDelay,] [Reason-code]) ReturnCode -- EndpointConfiguration( EndpointId, BearerInformation) ReturnCode -- NotificationRequest( EndpointId, [NotifiedEntity,] [RequestedEvents,] RequestIdentifier, [DigitMap,] [SignalRequests,] [QuarantineHandling,] [DetectEvents,] [encapsulated EndpointConfiguration]) ReturnCode -- Notify( EndpointId, [NotifiedEntity,] RequestIdentifier, ObservedEvents) ReturnCode, ConnectionId, [SpecificEndPointId,] [LocalConnectionDescriptor,] [SecondEndPointId,] [SecondConnectionId] --- CreateConnection(CallId, EndpointId, [NotifiedEntity,] [LocalConnectionOptions,] Mode, [{RemoteConnectionDescriptor | SecondEndpointId}, ] [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) ReturnCode, [LocalConnectionDescriptor] --- ModifyConnection(CallId, EndpointId, ConnectionId, [NotifiedEntity,] [LocalConnectionOptions,] [Mode,] [RemoteConnectionDescriptor,] [Encapsulated NotificationRequest,] [Encapsulated EndpointConfiguration]) My questions: 1) I am using Asterisk and a softphone called MGCP eyeP Phone. I was watching the traffic between them at Ethereal and observed that some of them have extra parameters. Example: CreateConnection has Request Identifier (X), that is not described on RFC 2705. Should I ignore or consider it? 2) There are some parameters that don't have identifier? I mean, Request Identifier is X:; Observed Events is O: ; Call ID is C: ... These are them: - Notified Entity - Remote Connection Descriptor - encapsulated Endpoint Configuration - Second Endpoint Id - encapsulated Notification Request 2.1) How can I identify them when they exist? 2.2) What means encapsulated
[Asterisk-Users] MGCP Transaction identifiers
I know this is not the most appropriated list to this, but I will try: Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method? I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP, CRCX and DLCX (by the call agent)use low values for it. I hope I could explain my doubt! Here are some examples: RSIP 18696 aaln/[EMAIL PROTECTED] MGCP 1.0 RQNT 8 aaln/[EMAIL PROTECTED] MGCP 1.0 AUEP 9 aaln/[EMAIL PROTECTED] MGCP 1.0 RSIP 27232 aaln/[EMAIL PROTECTED] MGCP 1.0 RQNT 3 aaln/[EMAIL PROTECTED] MGCP 1.0 NTFY 27219 aaln/[EMAIL PROTECTED] MGCP 1.0 CRCX 2 aaln/[EMAIL PROTECTED] MGCP 1.0 DLCX 26 aaln/[EMAIL PROTECTED] MGCP 1.0 Thanks, Leonardo J. Tramontina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Transaction identifiers
The criteria are published in RFC 3435, range is from 1 to 999,999,999. there is no requirement of starting from 1. Call agents may allocate certain ranges for certain groups of gateways. Asterisk (the call agent) simply increments the id numbers monotonically for each new request. Most likely the gateway (your phone) will do something like that too. My MGCP gateways are hardly ever turned off so the sequence numbers will get higher too. Leonardo J. Tramontina wrote: I know this is not the most appropriated list to this, but I will try: Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method? I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP, CRCX and DLCX (by the call agent) use low values for it. I hope I could explain my doubt! Here are some examples: RSIP 18696 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 RQNT 8 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 AUEP 9 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 RSIP 27232 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 RQNT 3 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 NTFY 27219 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 CRCX 2 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 DLCX 26 aaln/[EMAIL PROTECTED] mailto:aaln/[EMAIL PROTECTED] MGCP 1.0 Thanks, Leonardo J. Tramontina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Gateway
Any example for configuring T1 PRI with Asterisk using a Cisco 2600 series router? MGCP config? Do you Yahoo!? All your favorites on one personal page Try My Yahoo!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP
I havent found any recent information on this, but can Asterisk act as a MGCP UserAgent? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP
I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels - Original Message - From: Tim Jackson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 2:48 PM Subject: [Asterisk-Users] MGCP I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP
Any other ideas for interacting with an MGCP provider? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Tuesday, November 23, 2004 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MGCP I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels - Original Message - From: Tim Jackson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 2:48 PM Subject: [Asterisk-Users] MGCP I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP
I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels Any other ideas for interacting with an MGCP provider? You could, of course, connect an MGCP ATA to FXO port(s) or device(s). That solution degrades quality, increases delay, may have echo problems, etc. However, it's an easy way to get started, e.g. if you have a spare ATA-186 that you can load some MGCP firmware into. I am seeking a proper solution to the same problem, as my ISP in France, Free Telecom, bundles MGCP service at very aggressive rates (including free calls to fixed phones anywhere in France) with their ADSL service. I have looked at some SIP - MGCP and H.323 - MGCP gateways, but they only talk the Call Agent side of the protocol. If you have found a solution, please let me know. If not, perhaps we could work together to write one. One possibility is enhancing MGCP support in * to allow it to act as a User Agent. Another is a stand-alone script, e.g. in perl, that would do SIP - MGCP. I'd be open to other suggestions, too. Thanks, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP
The ILEC here is using VocalData for doing VoIP Centrex systems etc. My sales engineer preached SIP to me when he was talking about it, but I actually got a hold of an engineer today, and he told me they are using MGCP only for now. He seemed really interested in *, they are bringing out some demo units soon for us to beta-test for them and I told him I would show our * box to him. Maybe they might be interested in using MGCP with it, and would be willing to help out. I'll let you know. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stewart Nelson Sent: Tuesday, November 23, 2004 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels Any other ideas for interacting with an MGCP provider? You could, of course, connect an MGCP ATA to FXO port(s) or device(s). That solution degrades quality, increases delay, may have echo problems, etc. However, it's an easy way to get started, e.g. if you have a spare ATA-186 that you can load some MGCP firmware into. I am seeking a proper solution to the same problem, as my ISP in France, Free Telecom, bundles MGCP service at very aggressive rates (including free calls to fixed phones anywhere in France) with their ADSL service. I have looked at some SIP - MGCP and H.323 - MGCP gateways, but they only talk the Call Agent side of the protocol. If you have found a solution, please let me know. If not, perhaps we could work together to write one. One possibility is enhancing MGCP support in * to allow it to act as a User Agent. Another is a stand-alone script, e.g. in perl, that would do SIP - MGCP. I'd be open to other suggestions, too. Thanks, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP 1.0 NCS 1.0 on a motorola SBV4200
Hello all, Anyone have any experience using the Motorola SBV4200 cable modem with *. At my first try the CM was complaining about Incorrect Version. The problem was that the chan_mgcp in * is version MGCP 1.0 and the endpoint expects MGCP 1.0 NCS 1.0. Now from what I understand this NCS 1.0 is a packetcable extension of the MGCP protocol and is somewhat different. Now my question is wether anyone knows of a channel driver fully compatible with this packetcable extension ( at first glance the * -s chan_mgcp is not the best for this ) or any ideea what the differences are between the two protocols ? Kiss Karoly ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP problem
Hello, I have noticed a little problem in chan_mgcp.so. After a few unsuccessful attempts to call an endpoint using MGCP/aaln/[EMAIL PROTECTED] I have noticed the following on the system running asterisk using netstat. udp50524 0 XX.XX.85.XX:2427 0.0.0.0:* 2666/asterisk This line shows that there are 50524 bytes waiting in the Recvq of the udp socket of asterisk. And the transactions are all timing out. Snip Retransmitting #4 transaction 63 on [10.20.0.2] Retransmitting #4 transaction 64 on [10.20.0.2] Retransmitting #5 transaction 63 on [10.20.0.2] Retransmitting #5 transaction 64 on [10.20.0.2] Aug 24 14:49:14 WARNING[24601]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 63 on [10.20.0.2] Aug 24 14:49:14 WARNING[24601]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 64 on [10.20.0.2] Aug 24 14:49:14 NOTICE[24601]: chan_mgcp.c:2261 handle_response: Transaction 64 timed out Aug 24 14:49:14 NOTICE[24601]: chan_mgcp.c:2283 handle_response: Terminating on result 406 from aaln/[EMAIL PROTECTED] End snip. The system is GNU/Linux Debian 3.0r2 kernel 2.4.26. Asterisk CVS-HEAD-06/22/04-13:37:51 built by [EMAIL PROTECTED] on a i686 running Linux Any ideeas ??? Kiss Karoly ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and dialing out
I have recently found out that * is very strict about dialing out. If a number isn't listed in extensions.conf, good luck trying to dial it. I had to put in a line for each of our area codes with XXX's before I could dial local numbers. Anyway..now that I 'can' dial them, as soon as the other party picks up the phone I get a busy signal on my end. Also..just tried an IpPhone to IpPhone call. Works fine. However, when the other person hung up, I (on the swissvoice) got that busy signal again until I hung up. Any ideas on that busy signal plague? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP RFC3149 Swissvoice IP10
Title: MGCP RFC3149 Swissvoice IP10 I recently started using the Swissvoice IP10 phones with Asterisk. The only shortcoming I see is the fact that RFC3149 is missing from Asterisk which limits some of the Business uses. Thus I am a developer and considering making the necessary changes to including rfc3149 in Asterisk. Does anybody else out there have any use for RFC3149 with the IP10 phone or for other MGCP phones? What would be your uses for it? Any suggestions would greatly help in both developing a solution to fit both my needs and others. Thanks Chris Bennett
RE: [Asterisk-Users] MGCP Cisco ATA 186 Help
Hi, -Original Message- I've got it to work in the past. I've upgraded to SIP, seems to work better. Is there a reason you MUST have MGCP? Transfers work better with MGCP :-P Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Cisco ATA 186 Help
Yes. I use ata as test unit for MGCP system. No other available MGCP boxes. Next step will be cable modems with MTA. Duane Cox wrote: I've got it to work in the past. I've upgraded to SIP, seems to work better. Is there a reason you MUST have MGCP? Duane -- WBR - Dmitry Baranov Phone: +(372) 6 880 000 STV Internet Fax:+(372) 6 880 550 Valge 6 Mobile: +(372) 5 012 825 Tallinn, Estonia - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Cisco ATA 186 Help
Can you please sent me your mgcp.conf and screenshot from ATA config ? Leo Ann Boon wrote: My ATA with V3.0 firmware works fine. Check that test can be resolved by your DNS or is in /etc/hosts. You might just want to put the IP address directly. -- WBR - Dmitry Baranov Phone: +(372) 6 880 000 STV Internet Fax:+(372) 6 880 550 Valge 6 Mobile: +(372) 5 012 825 Tallinn, Estonia - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Cisco ATA 186 Help
Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line = aaln/2 line = aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485 __mgcp_xmit: mgcp_xmit returned -1: Address family not supported by protocol family *CLI Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 1 on [test] Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 2 on [test] Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 2 timed out Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 1 timed out *CLI Jul 31 16:05:44 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist mgcp debug MGCP Debugging Enabled *CLI MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.195.55:2427Verb: 'RSIP', Identifier: '1', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Jul 31 16:06:03 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist -- WBR - Dmitry Baranov Phone: +(372) 6 880 000 STV Internet Fax:+(372) 6 880 550 Valge 6 Mobile: +(372) 5 012 825 Tallinn, Estonia - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Cisco ATA 186 Help
I've got it to work in the past. I've upgraded to SIP, seems to work better. Is there a reason you MUST have MGCP? Duane - Original Message - From: Dmitri Baranov To: [EMAIL PROTECTED] Sent: Saturday, July 31, 2004 12:38 PM Subject: [Asterisk-Users] MGCP Cisco ATA 186 Help Does anybody has the expirience configuring Asterisk with Cisco ATA 186MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line = aaln/2 line = aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485 __mgcp_xmit:mgcp_xmit returned -1: Address family not supported by protocol family *CLI Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt:Maximum retries exceeded for transaction 1 on [test] Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximumretries exceeded for transaction 2 on [test] Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response:Transaction 2 timed out Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response:Transaction 1 timed out *CLI Jul 31 16:05:44 NOTICE[135449600]: chan_mgcp.c:1474find_subchannel: Gateway 'test' (and thus its endpoint '*') does notexist mgcp debug MGCP Debugging Enabled *CLI MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.195.55:2427Verb: 'RSIP', Identifier: '1', Endpoint:'[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Jul 31 16:06:03 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel:Gateway 'test' (and thus its endpoint '*') does not exist-- WBR - Dmitry Baranov Phone: +(372) 6 880 000 STV Internet Fax: +(372) 6 880 550 Valge 6 Mobile: +(372) 5 012 825 Tallinn, Estonia -___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Cisco ATA 186 Help
My ATA with V3.0 firmware works fine. Check that test can be resolved by your DNS or is in /etc/hosts. You might just want to put the IP address directly. Dmitri Baranov wrote: Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line = aaln/2 line = aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485 __mgcp_xmit: mgcp_xmit returned -1: Address family not supported by protocol family *CLI Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 1 on [test] Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 2 on [test] Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 2 timed out Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 1 timed out *CLI Jul 31 16:05:44 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist mgcp debug MGCP Debugging Enabled *CLI MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.195.55:2427Verb: 'RSIP', Identifier: '1', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Jul 31 16:06:03 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Caller ID
Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display Asterisk as the cid name, nothing for number. Here's what I have in my mgcp.conf for the device: [2084728800103] host = dynamic context = westcomllc line = aaln/1 callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] When placing outbound calls (out our pstn gateway), I always replace cid name/number w/the main number name of the company, so that direction it's not an issue -- just internal calls. Anyone seen this have ideas about what to do with it? Thanks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Caller ID
Try this: mgcp.conf [2084728800103]host = dynamiccontext = westcomllccallerid = "Jeremy Jones" 103nat = notransfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED] line = aaln/1 - Original Message - From: Jeremy Jones To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:13 AM Subject: [Asterisk-Users] MGCP Caller ID Good Morning,I'm having an issue with callerid display when calles are placed _from_an mgcp device (8x8 ata w/mgcp firmware). Internally, there are severaldifferent sip devices and one mgcp device. Calls from any of the sipdevices to any other device (sip or mgcp) have name/number displayedproperly by the called party's phone. Calls from the mgcp device to anyother device display "Asterisk" as the cid name, nothing for number.Here's what I have in my mgcp.conf for the device:[2084728800103]host = dynamiccontext = westcomllcline = aaln/1callerid = "Jeremy Jones" 103nat = notransfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED]When placing outbound calls (out our pstn gateway), I always replace cidname/number w/the main number name of the company, so that directionit's not an issue -- just internal calls. Anyone seen this have ideas about what to do with it?Thanks,Jeremy Jones___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Caller ID
Hi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Wednesday, July 28, 2004 7:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP Caller ID Try this: mgcp.conf [2084728800103] host = dynamic context = westcomllc callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1 Aha! Yup, that did the trick. So order matters there... Thanks, jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Caller ID
YES, because you could have an MGCP gateway device (more than one POTS line) ie. ours have 4 If so you would do something like this... [2084728800103]host = dynamiccontext = westcomllccallerid = "Jeremy Jones" 103nat = notransfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED]line = aaln/1 callerid = "Jeremy Jones #2" 104transfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED]line = aaln/2 ... etc... I do have a question for you though... I experimented with host=dynamic on the MGCP channel (we use MGCP here) I got it to work, but in this scenario, it was fatal, I'll explain and please tell me if you see the same thing. With host=dynamic, our MGCPend devicewould register with asterisk when powered up or when making the first call. All OK here, and asterisk would now remember (in memory) this registration, so any calls going back to the end device would be mapped appropriately. The fatality is that if asterisk is restarted, this "database of mapping" which was saved in memory; is now lost, so if a call came in and the end device was neverrebooted/restarted (to accomidate the asterisk restart) the mapping did not exist, as it was not saved in a "database" and the call would fail. So I switched back to host=ip.ip.ip.ip Do you see the same problem? Please let me know. Thanks Duane Cox - Original Message - From: Jeremy Jones To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:50 AM Subject: RE: [Asterisk-Users] MGCP Caller ID Hi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Wednesday, July 28, 2004 7:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP Caller ID Try this: mgcp.conf [2084728800103] host = dynamic context = westcomllc callerid = "Jeremy Jones" 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1Aha! Yup, that did the trick. So order matters there... Thanks,jeremy___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Caller ID
Hi Duane (et alia), YES, because you could have an MGCP gateway device (more than one POTS line) ie. ours have 4 If so you would do something like this... [2084728800103] host = dynamic context = westcomllc callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1 callerid = Jeremy Jones #2 104 transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/2 ... etc... I have, actually, a gazillion 4-port mgcp devices from a (recently-obtained-by-8x8) company called Centile that I've _never_ been able to get to work properly w/* -- maybe this info'll help me here... ...snip... The fatality is that if asterisk is restarted, this database of mapping which was saved in memory; is now lost, so if a call came in and the end device was never rebooted/restarted (to accomidate the asterisk restart) the mapping did not exist, as it was not saved in a database and the call would fail. So I switched back to host=ip.ip.ip.ip Yeah, that's an issue here, too. We primarily have sip devices, though, at all our customer sites, so it's only a problem with _my_ phone internally, which so far doesn't bother me (I hate talking on the phone, anyway). If I just pick up the handset connected to the mgcp device hangup, that magic mapping is re-created. I'd love to be able to deploy some of these things, though, for our customers I really wouldn't like all the maintainance involved in setting up static dhcp assignments for all these mgcp devices tying addresses to each mgcp endpoint in mgcp.conf. We have, as I mentioned, tons of these mgcp thingies lying around waiting for use. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and one-way audio
I have a D-Link 1120M MGCP Telephone Adapter with Asterisk using MGCP. Call setup / connects work ok. Can call from a fxs phone and can receive calls to the fxs phone. The problem is there is no audio to 1120M fxs phones. There is audio to the far end device (SIP and IAX). I have collected ether traces and do see RTP from the 1120M, but nothing from Asterisk to the 1120M. Any ideas? Here is my mgcp.conf [general] port = 2427 bindaddr = 192.168.1.30 disallow=all allow=ulaw [dlinkgw] host=192.168.1.32 context = local canreinvite = no line = aaln/2 line = aaln/1 Thanks, Phil
[Asterisk-Users] MGCP and call waiting, doesn't work.
Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick up the first.. Here is some debug info *CLI set verbose 10 -- Accepting call from '9003796075' to '9003790612' on channel 0/1, span 1 -- Executing Dial("Zap/1-1", "MGCP/aaln/[EMAIL PROTECTED]|30|tr") in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: -1, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/1-1 -- Accepting call from '2172026046' to '9003790612' on channel 0/2, span 1 -- Executing Dial("Zap/2-1", "MGCP/aaln/[EMAIL PROTECTED]|30|tr") in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: -1, dnd: 0, so: 1, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringingJun 29 09:46:58 NOTICE[98310]: chan_mgcp.c:2268 handle_response: Terminating on result 400 from aaln/[EMAIL PROTECTED] == No one is available to answer at this time -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] -- We didn't make one of the calls FLIPFLOP 0 and 1 on aaln/[EMAIL PROTECTED] -- MGCP Muting 0 on aaln/[EMAIL PROTECTED]Jun 29 09:47:08 WARNING[278545]: pbx.c:1892 ast_pbx_run: Timeout, but no rule 't' in context 'pstn-in' -- Hungup 'Zap/2-1' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' == Spawn extension (pstn-in, 9003790612, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here is my MGCP.conf [10.252.240.2]context=mainhost=10.252.240.2nat=nocallerid = "Kevin Osterbur" 9003790610callwaiting=yesthreewaycalling=notransfer=nocancallforward=nocanreinvite=noline = aaln/1context=mainhost=10.252.240.2nat=nocallerid = "Duane Cox" 9003790612callwaiting=yesthreewaycalling=notransfer=nocancallforward=nocanreinvite=noline = aaln/2 Any ideas?
[Asterisk-Users] mgcp/T1 interface/alternatives
Hi All, I just setup an asterisk system equiped to with a 4 FXO port board to interface it with my phone system (Inter-Tel). I realize that I'll be running our of ports once I start showing what can be done with it. My Inter-tel system supports MGCP connectivity to a gateway. Is there any planned development to make chan_mgcp work as a gateway? Another alternative, was to rig * and the phone system with T1 cards and hooking them up with a cat5 cord. I'm still confused about this one, is there any documentation about this set up?. I just want it to work like it's working now with the FXO board hooked up to analog ports in the phone system sorry for the ignorance I want to use * as a sip/iax gateway, for complex call routing, conference bridge, etc. In other words, to make it do what the phone system can't do, so connectivity with the phone system is essential. Are there any other alternatives? Regards, Renato A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Clients
Pro's advise me on Installation and Configuration of Asterisk to support Megaco (RFC 3015) Clients. Java _ Is your PC infected? Get a FREE online computer virus scan from McAfee® Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP RFC 3015
Friends i am new to Asterisk. I have bunch of Residential gateways Megaco RFC 3015 complaint. Has any one been successful in installing and running Asterisk in production environment with RGW RFC 3015 complaint clients ?? I am excited to see my RGW come to play if i can get Asterisk to work. Pros please advise me on Installation and Configuration of Asterisk to support Megaco Clients. Thank you, Java _ Learn to simplify your finances and your life in Streamline Your Life from MSN Money. http://special.msn.com/money/0405streamline.armx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP error dialing
Phillip group, I tried what you suggested and it did not work i included some more information for you to take a look at... i have got the MGCP working sort of for my asterisk server. My phone plugged into the dlink gateway does not ring when i call it. My sip phone does ring when i dial the extention. asterisk CLI shows its ringing correctly. I am using the dlink gateway whch has 2 ports in it. I have an extention for both ports in my extentions.conf. Asterisk appears to reconize that a phone is in there as it does not go to voice mail right away and it shows its dialing in the command line interface. I tried calling aaln/1 when no phone was in there and it went right to voice mail. I plug a phone in and it just says ringing. Below is my conf that i have now.Is there anything I need to configure in the Dlink gateway for this to work with asterisk? my gateway works fine and i use it normally for calls. I might have missed something very simple but I never tried this before so i am not sure... Dlink gateway Wan port to switch, Lan port no cable in it. line 1, normal phone plugged in *CLI show version Asterisk 0.9.0 built by [EMAIL PROTECTED] on a i686 running Linux *CLI extentions.conf [default] exten = 2002,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) exten = 2002,2,Hangup mgcp.conf [general] port=2427 ;bindaddr= [10.0.1.150] host=10.0.1.150 canreinvite=no context=default line = aaln/1 asterisk message output when you call the phone -- Executing Dial(SIP/2204-ac95, MGCP/aaln/[EMAIL PROTECTED]) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing asterisk output when you run asterisk -vvvgc chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found -- Allocating subchannel '0' on aaln/[EMAIL PROTECTED] -- Allocating subchannel '1' on aaln/[EMAIL PROTECTED] -- Added gateway '10.0.1.150' == MGCP Listening on 0.0.0.0:2427 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) Warning, flexibel rate not heavily tested! MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate when I run mgcp show endpoints *CLI mgcp show endpoints Gateway '10.0.1.150' at 10.0.1.150 (Static) -- 'aaln/[EMAIL PROTECTED] in 'default' is idle when I run mgcp audit endpoint aaln/[EMAIL PROTECTED] CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 6 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A,R,D,S,X,N,I,T,O,ES,VS,E,MD,M to 10.0.1.150:2427 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:417 mgcp_postrequest: Timeout waiting for response to message:1, lastouttime: 1085211476, now: 1085211613. Dumping pending queue May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 1 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 2 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 3 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 4 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 5 Any ideas? I followed wiki and any docs I can find on mgcp with the box.Everything else works... do i need to have something in bindaddr= for mgcp.conf? I marked that out. steven kalcevich Quoting Philipp von Klitzing [EMAIL PROTECTED]: Hi! I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? Do a mgcp show endpoints at the CLI and watch the output. May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01 WARNING[1251156800]: chan_mgcp.c:2608 mgcp_request: Unable to find MGCP endpoint 'aaln/[EMAIL PROTECTED]' mgcp.conf [dlinkgw] host=10.0.1.150 canreinvite=no context=default line = aaln/1 Change [dlinkgw] to [10.0.1.150], and the do a restart - depending on the Asterisk CVS version that you are using a reload or mgcp reload might not be sufficent/ might not work. See also: http://www.voip-info.org/wiki-Asterisk+config+mgcp.conf Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Steve Kalcevich Commercial Accounts Primus Telecommunications Canada Inc. Direct: 416-207-4613 Toll Free: 1-888-502-8380, ext. 8313 Fax: 1-800-861-3035 E
Re: [Asterisk-Users] MGCP error dialing
Hi! Below is my conf that i have now.Is there anything I need to configure in the Dlink gateway for this to work with asterisk? Here a few things you can try: - upgrade to CVS-HEAD (not 0.9.0) and see if things are different - issue a ngrep port 2727 to monitor what your dlink is sending - uncomment the bindaddr= statement Make sure you do a RESTART and not a RELOAD after any changes that are supposed to affect MGCP. If you continue to experience problems please open a bug report and include as much data as you can provide. In this case you might also want to try to go back to CVS HEAD of 03/05/04 00:50:56. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP error dialing
Hi! I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? Do a mgcp show endpoints at the CLI and watch the output. May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01 WARNING[1251156800]: chan_mgcp.c:2608 mgcp_request: Unable to find MGCP endpoint 'aaln/[EMAIL PROTECTED]' mgcp.conf [dlinkgw] host=10.0.1.150 canreinvite=no context=default line = aaln/1 Change [dlinkgw] to [10.0.1.150], and the do a restart - depending on the Asterisk CVS version that you are using a reload or mgcp reload might not be sufficent/ might not work. See also: http://www.voip-info.org/wiki-Asterisk+config+mgcp.conf Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP error dialing
I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? error I -- Executing Dial(SIP/2204-5dc2, MGCP/aaln/[EMAIL PROTECTED]) in new stack May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01 WARNING[1251156800]: chan_mgcp.c:2608 mgcp_request: Unable to find MGCP endpoint 'aaln/[EMAIL PROTECTED]' May 19 22:30:01 NOTICE[1251156800]: app_dial.c:536 dial_exec: Unable to create channel of type 'MGCP' == Everyone is busy at this time -- Executing VoiceMail(SIP/2204-5dc2, u2202) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/2' (language 'en') exten.conf setup with mgcp exten = 2202,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) exten = 2202,2,Voicemail(u2202) exten = 2202,3,Hangup mgcp.conf [general] port = 2427 bindaddr = 0.0.0.0 [dlinkgw] host=10.0.1.150 canreinvite=no context=default line = aaln/1 Regards, Steve Kalcevich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mgcp with busy tone
Hi there, ::: i have an mgcp phone (iph-90) using the sample mgcp.conf for it (the dlink section). i've tried both asterisk stable and development release but i'm getting the following error when i lift the receiver: . .. in stable branch: -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down while the phone is giving me busy tone . .. in development release: chan_mgcp.c:2227 handle_response: Terminating on result 502 from aaln/[EMAIL PROTECTED] while the phone after a short beep giving me busy tone too... any ideas? wiking ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP information
I have read over the archives and am still a little confused about mgcp support in asterisk. I realize the mgcp channel is server side only (but jump in and correct me if I got this wrong.) I have seen a few references to sipmgcp interteroperability in patches, so now I am wondering what the case is with support in the code. What I actually want to do is have asterisk act as the client for an mgcp pstn gateway service (primus.ca). Can asterisk do this ? If not is there any particular technical reason in the way of a channel driver for this, or simply not the demand up till now ? The reasons for this provider choice aren't pretty - just that they are the only provider with number migration capabilities in the CO where I need them. Jon Pounder Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP channel problem
Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. The identifier of first digit (2) is 7152 then asterisk received second digit (4), identifier 7153 and then asterisk received third digit... (2) with identifier 7152 so, asterisk dialed number 24254.. all debug is in attachment 1 headers, 0 lines Urgent handler MGCP read: NTFY 7152 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 2 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7152', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 7152 OK to 217.66.161.122:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8819 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 23f9c13c R: hu(N), hf(N), D/[0-9#*](N) S: ro to 217.66.161.122:2427 Urgent handler MGCP read: NTFY 7153 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 4 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7153', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 7153 OK to 217.66.161.122:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '4' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8820 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 23f9c13c R: hu(N), hf(N), D/[0-9#*](N) S: ro to 217.66.161.122:2427 Urgent handler MGCP read: NTFY 7152 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 2 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7152', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 7152 OK to 217.66.161.122:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8821 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 23f9c13c R: hu(N), hf(N), D/[0-9#*](N) S: ro to 217.66.161.122:2427 Urgent handler MGCP read: NTFY 7154 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 5 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7154', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] This problem is only while my colleague is downloading any data from internet. The voip gateway is on the same internet line as colleague's computer. I have these problemes everywhere with higher latence. Can I set digit report on my MGCP gateway to block mode ? I tried it, but no effect. I changed xgcp set_digit_report to 1 But it doesn't work :( My MGCP gateway always reports DTMF in comma separated. Can you help me please ? Thank you Vit Bohacek debug.txt Description: Binary data
[Asterisk-Users] MGCP Problem
Title: MGCP Problem Turns out this was a typo in my extensions.conf file all along. Many thanks to the person who pointed it out. The answer was staring me in the face the entire time, but I just couldn't see it. Apologies to all
Re: [Asterisk-Users] MGCP problem
Shouldn't the endpoint be MGCP/aaln/[EMAIL PROTECTED] instead of [EMAIL PROTECTED]? AFAIK, the MGCP RFC recommends aaln/# for analog lines. Can you audit the endpoint to check if you got the name right? Hope this helps. Juan J. Sierralta P. wrote: On Wed, 2004-05-05 at 13:45, Brad White wrote: This is a misunderstanding of the problem. There is not any traffic between the MG and MGC to look at because asterisk can't determine the endpoint, re: Are you using CVS head ? Attach your mgcp.conf and show us output of mgcp reload if you defined the MG correctly your * should be sending RSIP requests to the endpoint after the reload. BTW the message you saw doesn't mean that there is no traffic between both peers. -- Executing Dial(SIP/-0ccc, [EMAIL PROTECTED]) in new stack Apr 23 13:23:40 NOTICE[376847]: chan_mgcp.c:1457 find_subchannel: Gateway '66.17.13.240' (and thus its endpoint 'aaln2/') does not exist Apr 23 13:23:40 WARNING[376847]: chan_mgcp.c:3205 mgcp_request: Unable to find MGCP endpoint '[EMAIL PROTECTED]' Apr 23 13:23:40 NOTICE[376847]: app_dial.c:554 dial_exec: Unable to create channel of type 'MGCP' This is a sip client trying to call a mgcp endpoint. The very same mgcp endpoint gets sent dialtone from * and can also originate calls to either sip clients, or to * itself(the 500 demo for example). I'm not a programmer, but the problem as I see it is that * does place brackets around the ip address when the mgcp endpoint originates a call and when * audits the endpoint, but * does not place brackets around the ip address when the mgcp endpoint is the destination of a call. It could be bug that's why we need some MGCP traffic from the start for example the reload process and when the endpoint starts a call. What is the specific manufacter/model of your MG ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP: Current CVS works for you?
Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542 http://bugs.digium.com/bug_view_page.php?bug_id=881 and other MGCP related bugs/fixed. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP: Current CVS works for you?
Hi Philipp, I havn't tried latest mgcp code but I can say that chan_mgcp has serious problems with IP10S that are partially solved by my latest patch http://lists.digium.com/pipermail/asterisk-users/2004-March/041615.html I have received any feedback about it. Regards, Daniel ANDRE Philipp von Klitzing a écrit: Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542 http://bugs.digium.com/bug_view_page.php?bug_id=881 and other MGCP related bugs/fixed. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *** MGCP on the menu? Check today's special!
If you're using MGCP, we need your help. There's a patch in bugs.digium.com that needs testing by the community. Please spend some time testing and adding your comments to the bug tracker. The author writes: -- I'm trying to make work Asterisk against a Cisco IAD2431 with chan_mgcp. Since chan_mgcp assumes the Line package is the default which is not the usual with Cisco's GW I made a little patch which signals the package. Also the event cg is on Generic Media package so I changed cg to G/cg. I can make and receive calls now and I think it shouldn't hurt others MGCP endpoints. There are other problems, for example Cisco don't like how Asterisk signals more than one codec, since a:PCMU;PCMA should be used instead of a:PCMU, a:PCMA. And chan_mgcp always uses ulaw as the default even if a use allow in mgcp.conf. I'll try to make GSM work and some other stuff, like hook flash transfer and then send another patch since that could break interoperation with others mgcp GWs. - http://bugs.digium.com/bug_view_page.php?bug_id=693 Thank you for your help! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and IPH-90
Hi ::: i have an MGCP voip phone (IPH-90), but i couldn't get it work with asterisk. ::: i'm using asterisk 0.7.2 on openbsd 3.4 the config file and the debug infos are here: http://nostromo.jol.hu/asterisk/ so not to flood the mailing list. regards wiking ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP RELOAD function
Hello I was just wondering if anyone was working on the MGCP RELOAD functionality. Thanks, Duane Cox ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mgcp endpoint question
Title: mgcp endpoint question Hi, I'm testing an mgcp phone with * 0.72. While * accepts the endpoint gateway enclosed in [ ], the phone baulks when presented with the gateway domain not enclosed in [ ], see below. Is there anything I can add to my mgcp.conf to force the inclusion of the [ ]? I had a good old search through the archives and voip-info website, but find very little information on mgcp configurations on * MGCP read: NTFY 36364 00085D03009C/[EMAIL PROTECTED] MGCP 1.0 X: 0 O: hb 200 36364 OK RQNT 4522 00085D03009C/[EMAIL PROTECTED] MGCP 1.0 X: 612afe7a R: hd(N) S: vmwi(-) 500 4522 Endpoint is unknown - Gateway domain doesn't match ; ; MGCP Configuration for Asterisk ; [general] port = 2727 bindaddr = 10.50.100.39 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line = aaln/2 ;line = aaln/1 [10.50.100.200] host = 10.50.100.200 context = default line = 00085D03009C/1
[Asterisk-Users] MGCP media gateway
Hi, Anyone knows if there is a MGCP media gateway implementation for * ? The stock chan_mgcp.c seems to be a call agent implementation only and reflected in mgcp.conf as such. PK - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 09, 2004 3:39 PM Subject: Re: [Asterisk-Users] Dialing 800 numbers with VOIP Matt Lawson wrote: Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800 numbers. Are 800 numbers treated differently somehow? Or is there a business reason for disallowing them? It makes the ringing sound but never connects. You can call toll-free numbers via NuFone. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Problem.
Hi. Thanks for the tip. Now i'm able to recognize the endpoint MGCP. But i can not place a call. I'm attaching the debug from the asterisk and two configurations files (mgcp.conf and extension.conf). For my first test i just want to call the voice mail, that has the 112 extension. The debug from the Asterisk when is started. [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found -- Allocating subchannel '0' on aaln/0/[EMAIL PROTECTED] -- Allocating subchannel '1' on aaln/0/[EMAIL PROTECTED] -- Added gateway 'AP200' == MGCP Listening on 64.76.148.185:2427 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/0/[EMAIL PROTECTED] for hookstate Now the debug: MGCP_Debug.txt And the configuration files. mgcp.conf extensions.conf Any ideas? Thanks in advance. Ricardo Javier Martinez Ogalde Ingeniero de Desarrollo VOISS NET S.A. Cisco Certified Network Associate ,CCNA (CSCO 10643101) * : (56 2) 240 81 96 *: (56 2) 245 74 95 * : [EMAIL PROTECTED] *CLI mgcp debug MGCP Debugging Enabled *CLI MGCP read: ntfy 6013 aaln/0/[EMAIL PROTECTED] MGCP 1.0 X: 66a1316b O: l/hd from 64.76.148.186:2427Verb: 'ntfy', Identifier: '6013', Endpoint: 'aaln/0/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'ntfy' on aaln/0/[EMAIL PROTECTED] Transmitting: 200 6013 OK to 64.76.148.186:2427 -- Endpoint 'aaln/0/[EMAIL PROTECTED]' observed 'hd' -- Creating connection for aaln/0/[EMAIL PROTECTED] in cxmode: sendrecv callid: 6491271e35c6bea8 We're at 64.76.148.185 port 16786 Answering with capability 4 Posting Request: CRCX 2 aaln/0/[EMAIL PROTECTED] MGCP 1.0 C: 6491271e35c6bea8 L: p:20, a:PCMU M: sendrecv X: 35c6bea8 v=0 o=root 30682 30682 IN IP4 64.76.148.185 s=session c=IN IP4 64.76.148.185 t=0 0 m=audio 16786 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 64.76.148.186:2427 Jan 26 10:59:59 NOTICE[49159]: chan_mgcp.c:415 mgcp_postrequest: Timeout waiting for response to message:1, lastouttime: 1075125540, now: 1075125599. Dumping pending queue Jan 26 10:59:59 NOTICE[49159]: chan_mgcp.c:396 dump_queue: Removing message from aaln/0/[EMAIL PROTECTED] tansaction 1 -- MGCP Asked to indicate tone: dl on aaln/0/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 3 aaln/0/[EMAIL PROTECTED] MGCP 1.0 X: 35c6bea8 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 64.76.148.186:2427 -- MGCP mgcp_new(MGCP/aaln/0/[EMAIL PROTECTED]) created in state: Down MGCP read: 200 2 OK I: 0006BA1208 v=0 o=- 1075125262 1075125262 IN IP4 64.76.148.186 s=AddPac Gateway SDP c=IN IP4 64.76.148.186 t=0 0 m=audio 23012 RTP/AVP 0 from 64.76.148.186:2427Verb: '200', Identifier: '2', Endpoint: 'OK', Version: '(null)' 2 headers, 7 lines Capabilities: us - 4, them - 4, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 MGCP read: 200 3 OK from 64.76.148.186:2427Verb: '200', Identifier: '3', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines *CLI *CLI *CLI *CLI *CLI *CLI *CLI *CLI *CLI *CLI MGCP read: ntfy 6014 aaln/0/[EMAIL PROTECTED] MGCP 1.0 X: 35c6bea8 O: l/hu from 64.76.148.186:2427Verb: 'ntfy', Identifier: '6014', Endpoint: 'aaln/0/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'ntfy' on aaln/0/[EMAIL PROTECTED] Transmitting: 200 6014 OK to 64.76.148.186:2427 -- Endpoint 'aaln/0/[EMAIL PROTECTED]' observed 'hu' -- Modified aaln/0/[EMAIL PROTECTED] with new mode: recvonly on callid: 6491271e35c6bea8 Posting Request: MDCX 4 aaln/0/[EMAIL PROTECTED] MGCP 1.0 C: 6491271e35c6bea8 M: recvonly X: 35c6bea8 I: 0006BA1208 R: L/hd(N) to 64.76.148.186:2427 -- MGCP mgcp_hangup(MGCP/aaln/0/[EMAIL PROTECTED]) on aaln/0/[EMAIL PROTECTED] -- Delete connection 0006BA1208 aaln/0/[EMAIL PROTECTED] with new mode: recvonly on callid: 6491271e35c6bea8 Posting Request: DLCX 5 aaln/0/[EMAIL PROTECTED] MGCP 1.0 C: 6491271e35c6bea8 X: 35c6bea8 I: 0006BA1208 to 64.76.148.186:2427 -- MGCP Asked to indicate tone: on aaln/0/[EMAIL PROTECTED] in cxmode: recvonly Posting Request: RQNT 6 aaln/0/[EMAIL PROTECTED] MGCP 1.0 X: 35c6bea8 R: hd(N) to 64.76.148.186:2427 -- MGCP mgcp_hangup(MGCP/aaln/0/[EMAIL PROTECTED]) on aaln/0/[EMAIL PROTECTED] set vmwi(-) -- MGCP Asked to indicate tone: vmwi(-) on aaln/0/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 7 aaln/0/[EMAIL PROTECTED] MGCP 1.0 X: 35c6bea8 R: hd(N) S: vmwi(-) to 64.76.148.186:2427 MGCP read: 200 4 OK v=0 o=- 1075125262 1075125262 IN IP4 64.76.148.186 s=AddPac Gateway SDP c=IN IP4 64.76.148.186 t=0 0 m=audio 23012 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 0 2 4 8 18 from 64.76.148.186:2427Verb: '200', Identifier: '4', Endpoint: 'OK', Version: '(null)' 1 headers, 11 lines -- Creating connection for aaln/0/[EMAIL PROTECTED] in cxmode: inactive
RE: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
Hi, Yep, I got the latest firmware (and the next-to-latest, and the next-to-next-to-latest, and one earlier yet) for SIP. The first three (firmware versions 1228, 1227, and 1226) all have that password protected Advanced Configuration page. The fourth one I found (version ) is a bit more open. It appears mum's the word on that password, none of the requests others have made in, say, vonage forums for that top secret password have met with much success. I suppose I could try to brute force it, but I'm fairly lazy, I'd just as soon wait for someone to just blurt it out on accident in casual conversation. I did poke around in the binary files found the html stuff, as well as what appear to be clear-text default options for stuff one would find on the protected Advanced Configuration page. It may be possible for me to use one of the newer firmware versions, changing the options in the firmware binary before sticking it on the device, but I imagine I'd screw up a checksum somewhere if I edit the file directly. Haven't tried yet. So, next stop: sip with the firware. Thanks, Jeremy You can get the latest SIP firmware from Packet8's TFTP server at 4.42.235.170 file name current. Read more about it here http://web.packet8.net/download/ Only problems is, in this version the advanced configuration page with the SIP setup is password protected. If you look at the downloaded file, you can see all the HTML stuff for the configuration pages. It may be possible to figure out or remove the password protection. The other option is to load an older version of the SIP firmware in which the SIP page is not protected. I'm sure someone has a copy of it. By the way, do you have a copy of the MGCP firmware in case you want to go back to it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Problem.
Hi, -Original Message- from 64.76.148.186:2427Verb: 'ntfy', Identifier: '6001', Endpoint: 'aaln/0/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Jan 22 18:05:11 NOTICE[49159]: chan_mgcp.c:1102 find_subchannel: Gateway 'ap1' (and thus its endpoint 'aaln/0/0') does not exist Looks like your phone has been given a name ('ap1'). If your phone thinks it is called 'ap1' you should set that name in mgcp.conf too, otherwise it won't be recognised. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users