Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hello Gavin,
Sorry for so long time in my reply but I was very busy on other tasks.
I attached to this message my working test files for mgcp.
Best regards,
Daniel
Daniel ANDRE wrote:
Gavin Hamill a crit:
On Tue, 2003-11-04 at 10:14, Daniel ANDRE wrote:
Hullo Daniel :)
Can I request that you post the pertinent parts of your config to the
list, since I'm sure I'm not the only one who would benefit from a set
of known-working configs for these phones.
I will make some clean-up in my files and post them in a day or two. I
am not fully satisfied with my conf for now but it may help you.
Daniel
Personally, I'm on the verge of buying some SwissVoice handsets, simply
because the mix of feature-set, price, and build quality seems to be
untouchable.
The GrandStreams are about the same price, but the build quality looks
cheap and plastic - the IP10 actually looks like a business telephone.
Cheers,
Gavin.
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--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if stati=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=yes
[globals]
dan = sip/p-dan.phone.iris-tech.fr
swiss1 = mgcp/aaln/[EMAIL PROTECTED]
swiss2 = mgcp/aaln/[EMAIL PROTECTED]
;
;MACRO
;
[macro-apl1]
exten = s,1,Dial(${ARG1},30,Ttmr)
;#
[SIP]
;#
include = ent
[local]
include = ent
;
[default]
include = ent
[ent]
exten = 111,1,Macro(apl1,${swiss1})
exten = 112,1,Macro(apl1,${swiss2})
exten = 326,1,Macro(apl1,${dan})
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 192.168.10.254
[192.168.10.11]
host = 192.168.10.10
nat = no
disallow = all
allow = g711
allow = alaw
line = aaln/1
canreinvite = yes
[192.168.10.10]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
context=local
host = 192.168.10.10
nat=no
callerid = John 92
line = aaln/1
callgroup=0
cancallforward=yes
transfer=yes
line = aaln/1
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi Philipp,
Philipp von Klitzing a crit:
Hi!
I have an asterisk box with one GS101 register to it in SIP mode and an
IP10S in MGCP mode.
I can dial IP10S from my GS101 and everything seems fine.
But from my IP10S I can't dial any number (GS or anything else).
Is that GS specific, or does the problem also include other SIP UAs like
X-Lite, X-Pro, SJPhone etc?
No it does not depend on the phone called. I am trying to make an IP to
PSTN gateway and I can't dial any number with my IP10S
Did you try canreinvite=no?
yes. Here is my mgcp.conf:
---
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 192.168.10.254
[192.168.10.10]
threewaycalling=yes
transfer = yes
callwaiting = no
callwaitingcallerid = yes
host = 192.168.10.10
nat = no
disallow = all
allow = g711
allow = alaw
callerid = toto 111
line = aaln/1
-
You might try
Swissvoice support if the problem persists, they should have an intereset
to solve this ("rnc Info Lists" reported the same problem in a private
message earlier).
I will try it but I have understood that s/o has used IP10S with
success on this list so I have asked for it here before
In any case I'd be very interested to hear about your results, preferably
on this list. :-)
No problem
Regards,
Daniel
--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Daniel, the MGCP log you sent shows you sending the digits and asterisk receiving them, however after that either nothing happens (infinite digittimeout) or you cut the log short. Can you also send some console output with 'mgcp no debug' :-) It saves clutter. Maybe a peek at your extensions.conf might be usefull as well ? Also, can you tell us your phone's firmware ? (the IP10) I had one minor issue with the IP10 because of an older firmware version, a simple upgrade resolved it (by the way, in my case it was interpreting digits twice in some cases, i.e. dialling 326 would make asterisk think I was calling 33226) Best regards, Florian No it does not depend on the phone called. I am trying to make an IP to PSTN gateway and I can't dial any number with my IP10S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, your MGCP log shows that asterisk receives the digits ok, but the log is cut short so I don't see asterisk dealing with the digits. Can you tell us more about your extensions.conf ? One final thing I can think of, what firmware version does your IP10 have ? I had one minor issue with older firmware, an upgrade resolved it easily. Florian No it does not depend on the phone called. I am trying to make an IP to PSTN gateway and I can't dial any number with my IP10S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Daniel, the MGCP log you sent shows you sending the digits and asterisk receiving them, however after that either nothing happens (infinite digittimeout) or you cut the log short. Can you also send some console output with 'mgcp no debug' :-) It saves clutter. Maybe a peek at your extensions.conf might be usefull as well ? Also, can you tell us your phone's firmware ? (the IP10) I had one minor issue with the IP10 because of an older firmware version, a simple upgrade resolved it (by the way, in my case it was interpreting digits twice in some cases, i.e. dialling 326 would make asterisk think I was calling 33226) Best regards, Florian FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, -Original Message- FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. I now have: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 217. 114. 96. 205 Mac address 00:05:90:02:03:0d Protocol MGCP 1.0 Best regards Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, Florian Overkamp a crit: Hi, your MGCP log shows that asterisk receives the digits ok, but the log is cut short so I don't see asterisk dealing with the digits. Can you tell us more about your extensions.conf ? I have revisited my extensions.conf and seen that there were no context defined for my mgcp phones. So I have tried to define a proper context and define some dial plan for it in my extensions.conf. This didn't work. Next I have left the context blank in my mcgp.conf and modified the default dialplan in my extensions.conf and now my IP10S can dial out. Many thanks Florian for pointing this out. BTW is there some known issue with context keyword in chan_mgcp? One final thing I can think of, what firmware version does your IP10 have ? I had one minor issue with older firmware, an upgrade resolved it easily. Here is my info page: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 10. 10 Mac address 00:05:90:02:02:f0 Protocol MGCP 1.0 Daniel -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
Re: [Asterisk-Users] SwissVoice MGCP IP10S
On Tue, 2003-11-04 at 10:14, Daniel ANDRE wrote: Next I have left the context blank in my mcgp.conf and modified the default dialplan in my extensions.conf and now my IP10S can dial out. Hullo Daniel :) Can I request that you post the pertinent parts of your config to the list, since I'm sure I'm not the only one who would benefit from a set of known-working configs for these phones. Personally, I'm on the verge of buying some SwissVoice handsets, simply because the mix of feature-set, price, and build quality seems to be untouchable. The GrandStreams are about the same price, but the build quality looks cheap and plastic - the IP10 actually looks like a business telephone. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Florian Overkamp wrote: Hi, -Original Message- FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. I now have: Phone name Undefined why u did not define the name for this phone ? - it seams that name will be used as gw name in mgcp ... I'm about @[ip] Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 217. 114. 96. 205 Mac address 00:05:90:02:03:0d Protocol MGCP 1.0 Best regards Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, You seem to have nice recent firmware. I am not aware of any issues with the context configuration with MGCP, it all seems to work just fine for me. Strange... Best regards, Florian I have revisited my extensions.conf and seen that there were no context defined for my mgcp phones. So I have tried to define a proper context and define some dial plan for it in my extensions.conf. This didn't work. Next I have left the context blank in my mcgp.conf and modified the default dialplan in my extensions.conf and now my IP10S can dial out. Many thanks Florian for pointing this out. BTW is there some known issue with context keyword in chan_mgcp? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, -Original Message- Phone name Undefined why u did not define the name for this phone ? - it seams that name will be used as gw name in mgcp ... I'm about @[ip] Interesting. Actually I never defined it because it was not needed in my setup. Asterisk and the phone understand eachother just fine like this. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi! BTW is there some known issue with context keyword in chan_mgcp? For the sake of documentation: chan_mgcp doesn't reload configs on 'reload' http://bugs.digium.com/bug_view_page.php?bug_id=268 Since that bug has been resting there for some time now it might be good if anyone else that has made the same experience would add a comment to that bug to increase its weight...?! Secondly a question: From the two sources I come to understand that is *is* possible to run an MGCP phone behind NAT (opposed to what Florian stated earlier on this list)? My order of an ip10s is going out today, but maybe some of you MGCP folks can give this a try already now and report back? Finally I think someone should open a tiny bug note for a better sample mgcp.conf that comes with * - what do you think? Thanks, Philipp http://bugs.digium.com/bug_view_page.php?bug_id=129 add the option to prevent native bridge - canreinvite sometime I need to prevent to create native bridge in chan_mgcp [Quote from an archived message on this list:] After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This works: nat=yes line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 This doesn't: line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 nat=yes This makes sense if lines were treated as individual channels through NAT, but they aren't. NAT capability is dictated by the Gateway itself, and not each endpoint/subchannel. I hope this saves somebody some time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, -Original Message- Secondly a question: From the two sources I come to understand that is *is* possible to run an MGCP phone behind NAT (opposed to what Florian stated earlier on this list)? My order of an ip10s is going out today, but maybe some of you MGCP folks can give this a try already now and report back? Hmm, now that would be very welcome indeed Someone please prove me wrong on this account :-)) Finally I think someone should open a tiny bug note for a better sample mgcp.conf that comes with * - what do you think? Feel free to build one :-) [Quote from an archived message on this list:] After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This works: nat=yes line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 This doesn't: line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 nat=yes This makes sense if lines were treated as individual channels through NAT, but they aren't. NAT capability is dictated by the Gateway itself, and not each endpoint/subchannel. Hmmfun. I may try this, but not before the end of the week... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hello, I have experienced the 2 digits problem earlier. Here is my "old" configuration: Phone name Undefined Appli version IP10 M v0.2.0 (Build1) Boot version IP10 Boot v0.2.0 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 10. 11 Mac address 00:05:90:02:02:38 Protocol MGCP 1.0 With the new software version this pb disappeared: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 10. 10 Mac address 00:05:90:02:02:f0 Protocol MGCP 1.0 Regards, Daniel Florian Overkamp a crit: Hi, -Original Message- FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. I now have: Phone name Undefined Appli version IP10 M v0.3.0 (Build5) Boot version IP10 Boot v0.3.3 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 217. 114. 96. 205 Mac address 00:05:90:02:03:0d Protocol MGCP 1.0 Best regards Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hello all, I have a half working configuration: I have an asterisk box with one GS101 register to it in SIP mode and an IP10S in MGCP mode. I can dial IP10S from my GS101 and everything seems fine. But from my IP10S I can't dial any number (GS or anything else). All the version I use are the latest available Any Idea? Regards, Daniel Marian Danisek a écrit: rnc Info Lists wrote: Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. in sending you my mgcp.conf file, my ip10s mostly working fine... regards Marian ---mgcp.conf- [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Ji, -Original Message- I have an asterisk box with one GS101 register to it in SIP mode and an IP10S in MGCP mode. I can dial IP10S from my GS101 and everything seems fine. But from my IP10S I can't dial any number (GS or anything else). Is the callmanager setting on the IP10S correct ? (i.e. pointing to the asterisk box) Can you show 'mgcp debug' output ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, Florian Overkamp a crit: Ji, -Original Message- I have an asterisk box with one GS101 register to it in SIP mode and an IP10S in MGCP mode. I can dial IP10S from my GS101 and everything seems fine. But from my IP10S I can't dial any number (GS or anything else). Is the callmanager setting on the IP10S correct ? (i.e. pointing to the asterisk box) Yes it is Can you show 'mgcp debug' output ? I have attached the debug trace from dialling extension 326 Regards, Daniel Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 MGCP read: NTFY 6611 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 6746d764 O: hd from 192.168.10.10:2427Verb: 'NTFY', Identifier: '6611', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 6611 OK to 192.168.10.10:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- Creating connection for aaln/[EMAIL PROTECTED] in cxmode: sendrecv callid: 414339df6746d764 We're at 192.168.10.254 port 17648 Answering with capability 4 Posting Request: CRCX 8 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 414339df6746d764 L: p:20, a:PCMU M: sendrecv X: 6746d764 v=0 o=root 31799 31799 IN IP4 192.168.10.254 s=session c=IN IP4 192.168.10.254 t=0 0 m=audio 17648 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 192.168.10.10:2427 -- MGCP Asked to indicate tone: dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 9 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.10.10:2427 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down MGCP read: 200 8 OK I: 8 v=0 o=- 8 0 IN IP4 192.168.10.10 s=- c=IN IP4 192.168.10.10 b=AS:81 t=0 0 a=sendrecv m=audio 3 RTP/AVP 0 a=ptime:20 from 192.168.10.10:2427Verb: '200', Identifier: '8', Endpoint: 'OK', Version: '(null)' 2 headers, 9 lines Capabilities: us - 4, them - 4, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 MGCP read: 200 9 OK from 192.168.10.10:2427Verb: '200', Identifier: '9', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 6612 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 6746d764 O: 3 from 192.168.10.10:2427Verb: 'NTFY', Identifier: '6612', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 6612 OK to 192.168.10.10:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '3' -- MGCP Asked to indicate tone: dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 10 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.10.10:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 11 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) to 192.168.10.10:2427 -- MGCP mgcp_hangup(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Delete connection 8 aaln/[EMAIL PROTECTED] with new mode: sendrecv on callid: 414339df6746d764 Posting Request: DLCX 12 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 414339df6746d764 X: 6746d764 I: 8 to 192.168.10.10:2427 -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 13 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: ro to 192.168.10.10:2427 MGCP read: 200 10 OK from 192.168.10.10:2427Verb: '200', Identifier: '10', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: 200 11 OK from 192.168.10.10:2427Verb: '200', Identifier: '11', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: 250 12 OK P: PS=21,OS=3612,PR=0,OR=0,PL=0,JI=0,LA=0 from 192.168.10.10:2427Verb: '250', Identifier: '12', Endpoint: 'OK', Version: '(null)' 2 headers, 0 lines MGCP read: 200 13 OK from 192.168.10.10:2427Verb: '200', Identifier: '13', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 6613 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 6746d764 O: 2 from 192.168.10.10:2427Verb: 'NTFY', Identifier: '6613', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 6613 OK to 192.168.10.10:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 14 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 6746d764 R: hu(N), hf(N), D/[0-9#*](N) S: ro to 192.168.10.10:2427 MGCP read: 200 14 OK from 192.168.10.10:2427Verb: '200', Identifier:
RE: [Asterisk-Users] SwissVoice MGCP IP10S
At 23:49 31-10-2003 +0100, you wrote: Hi! MGCP works on IP basis, it has no userid's or passwords. Ouch - that means MGCP and NAT w/ dynamic IP (of the router) is a No-No? Correct. Use IAX :) Florian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, At 05:03 30-10-2003 +0300, you wrote: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK your device is not registered on * , [] name in mgcp.conf must be exactly as gw name in your case you have configured gw-name as 'ip10' in mgcp.conf but on your device it is '[192.168.0.5]' change it on device to ip10 or in * to [[192.168.0.5]] Actually, if we are talking about swissvoice phones then I must say I have not needed this. By the way, the exact gateway name is 192.168.0.5, without brackets (see log above). So this still does not explain why its not talking. I get the idea Asterisk is simply not writing anything back on the port to respond to the request. Are you up to date with CVS code ? Could you try and TCPDUMP to see what is communicated between Asterisk and the phone ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
rnc Info Lists wrote: Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. in sending you my mgcp.conf file, my ip10s mostly working fine... regards Marian ---mgcp.conf- [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi! in sending you my mgcp.conf file, my ip10s mostly working fine... Could you explain mostly in your sentence, and maybe - if you can - give quick overview of Grandstream vs. SwissVoice (except for the pending SIP implementation, of course)? Thanks, Philipp! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
rnc Info Lists wrote: Citeren rnc Info Lists [EMAIL PROTECTED]: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Were you able to configure the phones through their webinterface ? You could try entering 'mgcp debug' and then power up your phone to see if it registers at all... Yes, web config. of the phone works ok. The IP for the Asterisk server is in the call agent field and port 2427. The following comes on the Asterisk console at powerup. The items between the repeat. MGCP Show endpoints doesn't show anything. Evidently the phone isn't registered but not sure why since there doesn't seem to be a place to associate a userid or password. MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart ** from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users change the name of your gate from [192.168.0.5] to ip10 -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
rnc Info Lists wrote: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line = aaln/1 The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) exten = 3001,103,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi! MGCP works on IP basis, it has no userid's or passwords. Ouch - that means MGCP and NAT w/ dynamic IP (of the router) is a No-No? Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SwissVoice MGCP IP10S
I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line = aaln/1 The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,103,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Citeren rnc Info Lists [EMAIL PROTECTED]: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Were you able to configure the phones through their webinterface ? You could try entering 'mgcp debug' and then power up your phone to see if it registers at all... -- Met vriendelijke groet, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Citeren rnc Info Lists [EMAIL PROTECTED]: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Were you able to configure the phones through their webinterface ? You could try entering 'mgcp debug' and then power up your phone to see if it registers at all... Yes, web config. of the phone works ok. The IP for the Asterisk server is in the call agent field and port 2427. The following comes on the Asterisk console at powerup. The items between the repeat. MGCP Show endpoints doesn't show anything. Evidently the phone isn't registered but not sure why since there doesn't seem to be a place to associate a userid or password. MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart ** from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
