Re: [asterisk-users] Swissvoice IP10s setup
Are you using trixbos ? If yes have a look at the forums on www.trixbox.org - Original Message - From: Paul A Brown To: asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 9:18 PM Subject: [asterisk-users] Swissvoice IP10s setup Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul -- ___ --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] Swissvoice IP10S centralized phonebook
This is the information I got from Swissvoice support, I didn't tried yet, but if it can helps. How to use an external phone book IP10S phone supports access to Cisco Phone Book but not all functionalities. The IP10 uses his own interface to access to the Phone Book. If you want to connect to your remote phone book, you have to do the following actions: First, copy the URL under Search by name in a Web browser, for example: http://192.168.1.5/cisco/directory/searchDirectory.php You are going to have a XML file display in the Web Browser, like this one: CiscoIPPhoneInput TitleDirectory Search/Title PromptEnter search criteria/Prompt URLhttp://10.3.100.190:8080/ciscodirectory?action=listpage=0/URL InputItem DisplayNameFirst Name/DisplayName QueryStringParamfirstname/QueryStringParam InputFlagsA/InputFlags /InputItem InputItem DisplayNameLast Name/DisplayName QueryStringParamlastname/QueryStringParam InputFlagsA/InputFlags /InputItem InputItem DisplayNameNumber/DisplayName QueryStringParamnumber/QueryStringParam InputFlagsT/InputFlags /InputItem /CiscoIPPhoneInput Copy the information from the URL line (URLhttp://10.3.100.190:8080/ciscodirectory?action=listpage=0/URL). The easiest way to set the path in your phone is by the Web interface (but it could also be done by Telnet). Connect to your phone web server. Login and password are normally: admin Select Configure common phonebook. In Select phone book to use chose the value: Remote Then click on submit. File the box below with the URL you get previously: The IP address and the port number of the Phonebook server can be manually entered or synchronised with the Call Agent In our case, IP address: 10.3.100.190, Port number: 8080,Path: /ciscodirectory?action=listpage=0 Then click on submit. If you return to your phone and select the common phone book, it is normally connected to the remote one now. You can search by a name or if you put nothing and press on OK, it will return you the entire content of the remote phone book. More information about Cisco Phonebook management To manage remote phone book, you need a server. It could be one you developed by yourself or one include in your proxy server (not all of them include this feature). It must follow the Cisco implementation; you can have more information here: http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/3_1/index.htm Check for Cisco IP Phone Services Application Development Notes with Cisco CallManager 3.1. Igor Briski wrote: Anybody got any documentation/experience on the subject? I'm trying to get it working, but the documentation I have lacks any information on what should be installed on the server side. -- Igor Briški - [EMAIL PROTECTED] ___ --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 ___ --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] swissvoice
thomas DEILLON wrote: Hello, I have swissvoice phones and when i use one, a have in asterisk lines like: Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp -13691.-232125 have a idea ? Yes, Kevin said this earlier today: 2 wrote: i get lots of the below from friday 15.7.05 cvs as well ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... I will be looking into this issue later today. Doug ___ 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] swissvoice
Hi, -Original Message- I have swissvoice phones and when i use one, a have in asterisk lines like: Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp -13691.-232125 the swissvoice firmware is IP10 SP v1.0.0 (Build 11) and asterisk version is the cvs of 18 july 2005 (today). Swissvoice phones tend to have a few interesting side effects in their rtp timestamping, we have filed some issues on that. However, it would be fun to hear what the actual problem is you are experiencing :-) Best regards, 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
RE: [Asterisk-Users] Swissvoice IP 10S VoIP Telephone
-Original Message- From: Adrian Walker [mailto:[EMAIL PROTECTED] Sent: 09 December 2004 14:21 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Swissvoice IP 10S VoIP Telephone Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? http://www.definitive-edge.com/index-2Swis.htm This would be interesting except it appears to be a bit pricey. Am looking for a nice quality SIP phone that supports Message Waiting Indicator (Grandstream are too Fisherprice for my liking). If anyone has experience of it and also knows somewhere for a good price (bulk buying is fine). Cheers alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] Swissvoice IP 10S VoIP Telephone
Hi, -Original Message- Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? Yes. 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] Swissvoice IP 10S VoIP Telephone
Alex Barnes wrote: http://www.definitive-edge.com/index-2Swis.htm This would be interesting except it appears to be a bit pricey. Am looking for a nice quality SIP phone that supports Message Waiting Indicator (Grandstream are too Fisherprice for my liking). If anyone has experience of it and also knows somewhere for a good price (bulk buying is fine). Cheers alex Alex, The polycom IP 300 can be had for $115. Also, the new Sipura 841 (I think) should be shipping by now, and I ordered one of those for $84. I would like the Polycom more, because I don't have to deal with the Sipura Profile Compiler (which I still don't have). They are even worse than Polycom when it comes to this issue. -- Kristian Kielhofner ___ 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] Swissvoice IP10S opinions?
Any luck with these phones and their SIP firmware? I just changed one of my IP10s to use SIP and I can't get it to work with Asterisk (it doesn't register at all). Regards, Alejandro. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Saturday, October 30, 2004 6:09 AM To: JB Hewit; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Swissvoice IP10S opinions? Hi, On Sat, 2004-10-30 at 02:50, JB Hewit wrote: Hi, I'm looking at trying out an IP10S with Asterisk. I'll be recieving a single unit next week to try out and see what she can do. It seems to be comparable to a Snom190, but I don't seem to find much detail online about it with Asterisk. Is anyone out there using these phones? Any quirks, reviews, goodness, badness about them? Use the MGCP firmware, its a lot more mature than their new SIP version. I'm currently rolling out several hundreds of them and they make excellent 'standard issue' desk phones. Asterisk could get a bit more work in the Business package support, but thats all fancy. There was an issue with RTP, because the IP10 can only deal with 1 RTP stream at a time, which is why I asked Digium to implement the singlepath option :-) 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 ___ 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] Swissvoice IP10S opinions?
Hi, On Sat, 2004-10-30 at 02:50, JB Hewit wrote: Hi, I'm looking at trying out an IP10S with Asterisk. I'll be recieving a single unit next week to try out and see what she can do. It seems to be comparable to a Snom190, but I don't seem to find much detail online about it with Asterisk. Is anyone out there using these phones? Any quirks, reviews, goodness, badness about them? Use the MGCP firmware, its a lot more mature than their new SIP version. I'm currently rolling out several hundreds of them and they make excellent 'standard issue' desk phones. Asterisk could get a bit more work in the Business package support, but thats all fancy. There was an issue with RTP, because the IP10 can only deal with 1 RTP stream at a time, which is why I asked Digium to implement the singlepath option :-) 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] Swissvoice IP10S and RTP Port Operation
Hi Matthew, -Original Message- What is even better is that this coincides with the Terminating on result 502 from [EMAIL PROTECTED] error I get in * So, I am guessing these are related. Any help here would be greatly appreciated. I am so close to getting this phone working in MGCP mode. Did you post your phone config to the list ? (mirror pages or something) I don't really understand why it's so hard for you where my phones did basic functions (calling/being called) almost right out of the box. 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] Swissvoice ip10s
Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ set features new 2 Operator NOINFO NOCONF FALSE extension of your secretary And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). 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] Swissvoice ip10s
Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Jun 24, 2004, at 12:33 AM, Florian Overkamp wrote: Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ set features new 2 Operator NOINFO NOCONF FALSE extension of your secretary> And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). 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] Swissvoice ip10s
Hi, -Original Message- Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Ouch! Can you check if it is still fetching any config files from your FTP-server at boot ? Might be your configs are corrupted somehow. If it is not even doing that, you might just have to ship it back to SwissVoice and have them fix it :-P Best regards, 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] Swissvoice ip10s
Yep I stayed and was able to get through to their ip-phone support in france. And with me only knowing english and the guy on the other end speaking broken english we kinda hashed out that it was a bad stick of flash ram in the phone. Communitech the USA provider for the phone is overnighting me a new one. AND emailing me the sip firmware for the mgcp phones. Thanks, Matt Hohman - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 9:24 AM Subject: RE: [Asterisk-Users] Swissvoice ip10s Hi, -Original Message- Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Ouch! Can you check if it is still fetching any config files from your FTP-server at boot ? Might be your configs are corrupted somehow. If it is not even doing that, you might just have to ship it back to SwissVoice and have them fix it :-P Best regards, 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 ___ 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] Swissvoice ip10s
Ah nice, Let me know what SIP version you get, if it's any more recent than the one I have, I'd love to get a copy ;-) (There are some issues in my version that make the phone rather useless) Florian -Original Message- Yep I stayed and was able to get through to their ip-phone support in france. And with me only knowing english and the guy on the other end speaking broken english we kinda hashed out that it was a bad stick of flash ram in the phone. Communitech the USA provider for the phone is overnighting me a new one. AND emailing me the sip firmware for the mgcp phones. ___ 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] SwissVoice IP phones
On Thu, 2004-05-20 at 23:10, John Vogel wrote: Has anybody used these with Asterisk? http://www.google.com/search?hl=enie=UTF-8q=site%3Alists.digium.com+SwissVoicebtnG=Google+Search -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] Swissvoice ip10: No 3-way-calling! (MGCP)
Hi, -Original Message- IP10S have not the capabilities to mix by itself 2 RTP flows, that why it refuses the conference mode. Most often, with the different partner we have, the conference capability is managed by a MCU on the network, then on the phone side, it's transparent. Yep, that's too bad. But how about that transfer issue that's pending ? Hope that can be solved... 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] SwissVoice IP10S not able to dial calls
Hi! I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. Which verasion of Asterisk are you using? Please do check the known MGCP bugs, especially 881: http://bugs.digium.com/bug_view_page.php?bug_id=881 Also don't forget to provide info about your ip10 firmware version. Finally: Did you put the ip10 into the right context so that it actually has the required access rights to dial 7999? Cheers, Philipp I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '9' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed -- MGCP handle_request(aaln/[EMAIL PROTECTED]) set vmwi(-) Here are my configuration files: MGCP.conf === [10.1.24.112] context=local host=10.1.24.112 callerid = Brad Chilton 7726 callgroup=0,2-5 canreinvite=no pickupgroup=0,1 nat=no threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer ;callwaiting=yes ; this might be a cause of trouble for ip10s ;cancallforward=yes line = aaln/1 EXTENSIONS.conf === exten = 7726,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) ___ ___ 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] swissvoice ip10s
Hi! does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. Never tried, no clue. But I can tell you that newer ip10 firmware and latest head CVS (yesterday) don't play together at all - see bug 881. Appli version IP10 M v1.0.0 (Build3) Boot version IP10 Boot v0.3.6 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) Protocol MGCP 1.0 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] 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
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
