[Asterisk-Users] IAX2 with g729 ATA Device
I am trying to find an ATA that will provice IAX2 and g729. I have not had much luck, I am hoping someone here might have some ideas. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] SIP Soft Video phone for Asterisk usage
Try Eyebeam from xten.com. I usethe phone part but haven't tried the video yet. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Ronald Wiplinger wrote: I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tying together two Asterisk servers
I am trying to tie together two asterisk servers. I keep getting an
error is clearly indicating I have missed something in the configuration
but I am just not sure what.
Server A has connection to the PSTN and Server B has SIP and other
phones connected but no PSTN.
The relevant configurations and error messages are as follows:
Server A:
May 25 13:52:34 NOTICE[4362]: chan_iax2.c:5448 socket_read: Rejected
connect attempt from 66.230.104.113
iax.conf
[finitetech]
type=peer
host=dynamic
secret=password
context=iaxtrunks
allow=all
trunk=yes
extensions.conf
[iaxtrunks]
exten = _1NXXNXX,1,Dial(Zap/R1/9072673700,20,D(9073372860${EXTEN:0}))
Server B:
May 25 12:53:19 WARNING[9437]: chan_iax2.c:5546 socket_read: Call
rejected by 12.166.160.13: No authority found
iax.conf
register = finitetech:[EMAIL PROTECTED]
extensions.conf
exten =
_81NXXNXX,1,Dial(IAX2/finitetech:[EMAIL PROTECTED]/D(${EXTEN:[EMAIL PROTECTED])
--
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
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[Asterisk-Users] Port 6057 blocked on firewall
When using Xten's Eyebeam software I am noticing that I get a blocked port 5067 on my firewall. The source port obviously varies but the 6057 seems to be consistent. I have done some looking and can find any reference to what may be happening here. I am guessing I need to modify some packet filters but I would like to make sure I understand this so I can open the right port ranges. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] 7960 multi-line configuration
You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Just curious, why do you need the same line to appear on multiple line buttons. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Kristof Hardy wrote: Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should I don't think that is possible, at least not the way one thinks it would work. I have also done some reading on this, maybe this thread gives a solution: http://lists.digium.com/pipermail/asterisk-users/2004-March/039271.html But, I am also curious on how other people have solved this, especially with using AMP for example. Cheers.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 multi-line configuration
You can configure multiple button to use multiple extensions. You do this in the phones configure file, here is a quick example. In this example the phone is the receptionist for two companies and it also has an individual line appearance for the the receptionist DID line. Keep in mind that if you configure the voice mail box in the sip.conf fill then you will see a voice mail indicator for each line as well. You will need to create entries in the sip.conf file for each extension as well as dial plan entires in extensions.conf but then all should be well # SIP Configuration Generic File # Line 1 line1_name: CRV_Reception line1_authname: "crv_reception" line1_password: "crv_reception" # Line 2 line2_name: "KP_Reception" line2_authname: "kp_reception" line2_password: "kp_reception" # Line 3 line3_name: "Colleen" line3_authname: "colleen" line3_password: "colleen" # Line 4 line4_name: "Line 4" line4_authname: "UNPROVISIONED" line4_password: "UNPROVISIONED" # Line 5 line5_name: "Line 5" line5_authname: "UNPROVISIONED" line5_password: "UNPROVISIONED" # Line 6 line6_name: "Line 6 line6_authname: "UNPROVISIONED" line6_password: "UNPROVISIONED" Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Matthew Boehm wrote: Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf dial plan
Here isa quick example, this rings the reception only for 10s and then
rings reception and another phone for 15s and then finally the voicemail.
; Reception
exten = 8900,1,Dial(SIP/crv_reception,10,Ttr); SIP - Reception
exten = 8900,2,Dial(SIP/crv_receptionSIP/gayle,15,Ttr)
exten = 8900,3,Voicemail(u8900)
exten = 8900,4,Voicemail(b8900)
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
Georg P. Israel wrote:
Dear Asterisk users,
I was wondering if anybody can tell me how to define a dial scheeme such
that an incomming all first rings for e.g. 20 seconds on one set of
phones and then after this time extends it's range onto a bigger set of
phones.
Basically, this is easy,
I can do this in the extensions.con with
[ISDN-in]
exten= 6201030,1,setcallerid(${CALLERID} ${CALLERID}|a)
exten= 6201030,2,dial,${UserGroup1}|20|t
exten= 6201030,3,dial,${UserGroup1UserGroup2}|60|t
exten= 6201030,4,Voicemail2(u6201030)
exten= 6201030,5,hangup
exten= 6201030,302,Voicemail2(b6201030)
But here is on major problem,
in step 2, after 20 seconds, the call on the phones in Group1 will be
terminated and then restarted in the bigger group (Group1Group2).
The problem with this is, during the transition is a time gap of a view
seconds on the phones from Group1. That means, if I lift up the head set
during this gape, then I can loos the calls on those phones.
Hence, I was wondering if I can set the dial proceadure such, that I
have the calls for 80 seconds on the phone Group1, and after 20 seconds
additionally on the phone Group2 without any interruption of the ringing
on the other phones.
Best regards
Georg P. Israel
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Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
I have tried several, dlink doesn't seem to have the same issue and a more intelligent firewall is not having any problems. We are working with the Sipura 1001 and 2000 units on this issue. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri="sip:asterisk.mydomain.com", ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri="sip 66.x.x.166:5060", ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Help compiling zaptel in Debian
I have compile Asterisk many times on Woody without any problems. I am not sure about the error your are receiving but some things that you might want to watch for. * You need to recompile the kernel with the source for Woody I think this is currently 2.4.18 I noticed you have 2.4.20. * Use the kernel definition distributed with Woody, don't make this up as you go, or things will get ugly fast every time I think I know better I end up recompiling after it doesn't work. * Get the tar ball distribution of Asterisk, Zaptel and libpri, don't use the CVS. I don't know CVS that well and I have always had trouble getting the right version of the code. The tar balls are a nice package and they are stable releases. * For the time being avoid the Debian packages, recently they got the libpri out of sync and we had to compile things manually to bring our system back up after everything came tumbling down. * Don't do anything fancy, Woody is stable, asterisk and the support software will compile if you do everything just right ;) * All else fails you can use Sarge but the process is the same as are the pit falls. While all of our development systems are running Sarge now, Sarge still has a large number of updates that you need to keep up with and this can be frustrating. Finally, if you are still having trouble, we build and compile these systems regularly for customers and we can help you if you need the assistance. Good luck Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tzafrir Cohen wrote: On Mon, Apr 18, 2005 at 04:28:26PM +0200, Manuel Casal wrote: During the zaptel configuration at the end of it there is this error: What version of zaptel? post-install tor2 /sbin/ztcfg post-install wcusb /sbin/ztcfg post-install wcfxo /sbin/ztcfg post-install ztdynamic /sbin/ztcfg post-install ztd-eth /sbin/ztcfg post-install wct1xxp /sbin/ztcfg post-install wct4xxp /sbin/ztcfg post-install wcte11xp /sbin/ztcfg if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.20-686/misc/ztd-eth.o [ `id -u` = 0 ] /sbin/depmod -a || : depmod: *** Unresolved symbols in /lib/modules/2.4.20-686/misc/ztd-eth.o What are those symbols? depmod -a -e Only a problem with ztd-eth? Any change that you didn't build it this time? Look at the dates of files in /lib/modules/2.4.20-686/ [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf I have previously made the make oldconfig and the make dep ... I'm using debian with 2.4.20 kernel... Debian stable , that is, right? /me wonders what would it take to freshen the backport of asterisk. Any volunteers? ___ 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] Asterisk timer on Digium's TDM cards?
You might want to confirm this but I think the basic analog cards have the same functionality and they are very inexpensive. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Boris Bakchiev wrote: Hi, Wondering if there is a timer provided on TDM cards? I don't have use for TE110P and it seems expensive just to get it for timer function. I do have ztdummy running but it is hovering on 99.975586% and I'm not sure if this is good enough or not. Any info is appreciated. This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller in meetme room quiet (low level?)
I have encountered a frustrating problem with the meetme rooms and calls entering the system on the Digium analog cards. The typical scenario is: Callers on SIP phones, X-lite, Eyebeam, Cisco 7960, IAXy Callers entering the system from the PSTN via the digium Analog card (TDM400P) In the meetme room the SIP connections can all hear each other loud and clear. The PSTN people can hear everyone just fine but the SIP users have trouble hearing the analog PSTN connections. The quality of the PSTN connections is good but the levels (volume) are low. There seems to be some variability in the apparent level depending on the SIP device that is listening. The IAXy doesn't seem to have any problems with the level. I have made some adjustments to the levels on the analog card but when i go to far the echo canceling converges slowly and is very annoying at the beginning of the call. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Polycom IP 3000 configuration
I have set Asterisk as a gateway on the Polycom and set gatekeeper to
"No"
So to dial on the Polycom I would then dial (0+the number). No way to
just dial directly without the 0?
The other side of this is how do I dial "to" the Polycom, I have tried
everything that I can think of for the "exten" definition and nothing
seems to work.
I did this setup via the web interface so I can't test until Monday.
Thanks
Tim Courcy wrote:
You need to
set the asterisk as a gateway
in the polycom.. then to dial out. Lets say you set the * as GW 0 on
the
polycom you would dial 0*{exten} in order to dial through a gw on the
ip3000
you have to use the prefix for the gateway. So 0* for GW 0 and 1* for
GW 1
Hope this
helps if you need more info mail
me off list.
Thanks
Tim
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Scott Henderson
Sent: Saturday,
February 12, 2005
7:12 PM
To: Asterisk Users
Mailing List -
Non-Commercial Discussion
Subject: Re:
[Asterisk-Users]
Polycom IP 3000 configuration
I see that typo I made for
this suggestion, but the
real problem is that the system doesn't seem to register with Asterisk.
I can't dial out or even if I fix the error in my config will I be able
to dial
the extension.
This phone just doesn't seem to want to work with Asterisk. I have
found
some old posts where people got this phone to work but they never post
the
solution so i am hopeful someone has the answer.
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
harry gaillac wrote:
hello
try: exten = 8908,1,Dial(h323/8908,20,Ttr) !
harry
--- Scott Henderson [EMAIL PROTECTED] a crit :
I am trying to add a Polycom IP 3000 to our Asterisk
system and am not
getting anywhere.
h323.conf
[8908]
type=friend
host=192.168.104.25
secret=polycom
context=crv-default
callerid="Conference Room Polycom"
extensions.conf
exten = 8908,1,Dial(h323/polycom,20,Ttr)
; Polycom
exten = 8908,2,Hangup
I have tried setting the Asterisk system as both
gatekeeper and gateway
in the polycom config.
To date nothing seems to work and Polycom is now on
a week return a
support call to the reseller that sold us the unit.
--
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time:
http://www.worldtimeserver.com/time.asp?locationid=US-AK
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Re: [Asterisk-Users] Polycom IP 3000 configuration
I see that typo I made for this suggestion, but the real problem is that the system doesn't seem to register with Asterisk. I can't dial out or even if I fix the error in my config will I be able to dial the extension. This phone just doesn't seem to want to work with Asterisk. I have found some old posts where people got this phone to work but they never post the solution so i am hopeful someone has the answer. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK harry gaillac wrote: hello try: exten = 8908,1,Dial(h323/8908,20,Ttr) ! harry --- Scott Henderson [EMAIL PROTECTED] a crit : I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten = 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten = 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper and gateway in the polycom config. To date nothing seems to work and Polycom is now on a week return a support call to the reseller that sold us the unit. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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 Dcouvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Crez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 3000 configuration
I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid=Conference Room Polycom extensions.conf exten = 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten = 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper and gateway in the polycom config. To date nothing seems to work and Polycom is now on a week return a support call to the reseller that sold us the unit. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Connecting Phone To Asterisk
How does the phone know which server's address, did you set this with DHCP or directly into the phone. If you already set this information can you verify it in some way. If you are not seeing anything on the server then the phone is most likely not talking to it for some reason. Daniel Joos wrote: Here is the discovery that I have made. * I can Ping my phone w/o any problems. * Near as I can tell, I have set up the phone correctly in sip.conf * When I set up sip debug and/or sip debug ip and I reboot my phone, I do not see any activity from the server from my phone. Username Secret Accountcode Def.Context ACL NAT 1202password default No No 1201password default No No Name/usernameHost DynNat ACL Mask Port Status 1202/1202 (Unspecified) 0.0.0.05060 Unmonitored 1201/1201 192.168.0.10 255.255.255.255 5060 Unmonitored ___ 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 -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Connecting Sip phone to asterisk.
Try debugging sip with debug sip at the CLI. This will provide you with more detail on what is going wrong. Scott [EMAIL PROTECTED] wrote: I am having a major dillema here, I have been trying to get my sip phone (hard phone) to communicate with the asterisk server. Below is my configuration: sip.conf [1201] type=friend username=1201 secret=password mailbox=1201 host=dynamic [1202] type=friend username=1202 secret=password mailbox=1202 host=dynamic extensions.conf exten = 1201,1,Dial(SIP/1201,9,rt) exten = 1201,n,Voicemail(u1201) exten = 1201,s+1,Hangup exten = 1202,1,Dial(SIP/1202,9,rt) exten = 1202,n,Voicemail(u1202) exten = 1202,s+1,Hangup Anyone have any ideas on if there is something that is missing? Also, the model of the phone is KE1020A ___ 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 -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Re: Connecting Sip phone to asterisk.
I am guessing that the core of your problem here is that the phone isn't registering with the server (a statement of the obvious). Anyway I just went through this on a project and if you study the debugs, you will see where the registration falls down. Mine was a series of typos on both the phones configuration and the .conf files. It took look at the debugs repeatedly to track down all the little errors that were between my and the registration. Also take a look as sip show peers and sip show users to see what you can see there. Scott [EMAIL PROTECTED] wrote: The phone is configured as: IP Phone Number: 1201 Username: 1201 Password: password Service Address: 192.168.0.104 Sip.conf is configured as: [1201] type=friend username=1201 secret=password mailbox=1201 host=192.168.0.99 To keep the redundant data down, here is what the sip debug shows: Retransmitting #5 (no NAT): INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK6a747077 From: 1202 sip:[EMAIL PROTECTED];tag=as166994fa To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 08 Jan 2005 20:22:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 v=0 o=root 6552 6552 IN IP4 192.168.0.104 s=session c=IN IP4 192.168.0.104 t=0 0 m=audio 11670 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.99:5060 Jan 8 12:22:25 WARNING[6552]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Destroying call '[EMAIL PROTECTED]' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Re: Connecting Sip phone to asterisk.
One more thing, what phones are you using? [EMAIL PROTECTED] wrote: The phone is configured as: IP Phone Number: 1201 Username: 1201 Password: password Service Address: 192.168.0.104 Sip.conf is configured as: [1201] type=friend username=1201 secret=password mailbox=1201 host=192.168.0.99 To keep the redundant data down, here is what the sip debug shows: Retransmitting #5 (no NAT): INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK6a747077 From: 1202 sip:[EMAIL PROTECTED];tag=as166994fa To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 08 Jan 2005 20:22:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 v=0 o=root 6552 6552 IN IP4 192.168.0.104 s=session c=IN IP4 192.168.0.104 t=0 0 m=audio 11670 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.99:5060 Jan 8 12:22:25 WARNING[6552]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Destroying call '[EMAIL PROTECTED]' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Multiple lines on Cisco 7960
I did set these to the correct poxy serveras well in the SIPDefault.cnf file. This is very frustrating problem, I have ready dozens of posts that refer to how to set this up and I see mto have followed all the suggestions. I had not looked at the phones settings yet, thanks for the suggestion. The setting indicate that there is no configuration on the second line it is listed as UNPROVISIONED Scott Nathan Alberti wrote: Do you have: # Proxy Server proxy1_address: x.x.x.x proxy2_address: x.x.x.x Unsure if this is required, does your phone list the correct server ? (settings | 4 | 2 | 6) Nathan. Scott Henderson wrote: I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple sip debug and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend host=dynamic username=scott1 secret=scott1 context=default mailbox=6101 callerid=Scott Henderson 1 macaddress.cnf # Line 1 line1_name: Scott line1_authname: scottline1_password: scott # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 sip debug output from resetting the phone: Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 Expires: 3600 10 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0045611f Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=scott,realm=asterisk,uri=sip:192.168.17.13,response=7b9f392d15161ef76ae35f283e876497,nonce=0045611f,algorithm=md5 Content-Length: 0 Expires: 3600 11 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 Date: Fri, 07 Jan 2005 02:56:25 GMT Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Date: Fri, 07 Jan 2005 02:56:26 GMT CSeq: 102 NOTIFY Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' argon*CLI The result of this configuration is that I always get the first line line_1 but never the second
Re: [Asterisk-Users] Message light on 7960 or in this case no message light
I think the issue is the context specification. In this application I had two contexts in voicemail.conf that were not default. I have modified the sip.conf as suggested. Scott Nathan Alberti wrote: Ensure you have mailbox= in sip.conf, you must also make sure in voicemail.conf the mailbox declarations are under the [default] context. If this is not the case you need to specify the context. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20mailbox i.e. # voicemail.conf # [admin] 4060 = 4060,fred,[EMAIL PROTECTED] [sales] 4061 = 4061,Sales Team,[EMAIL PROTECTED],,delete=yes # sip.conf # [4060] .. [EMAIL PROTECTED] [4061] .. [EMAIL PROTECTED] Scott Henderson wrote: I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip chatter is almost none existent and the sip chatter on the first system that works is quite regular. There is a version difference as follows: The system that is working is: Asterisk 1.0.1 built by [EMAIL PROTECTED] on a i686 running Linux The system that isn't working is: Asterisk 1.0.2 built by [EMAIL PROTECTED] on a i686 running Linux I have reviewed everything I can think of but now message lights and the chatter that seems to have the Message information doesn't seem to be occurring on the system that isn't work like it is on the working system. ___ 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 -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing an extension on multiple phones
I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done or is there a better way. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Multiple lines on Cisco 7960
3.1 ### # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into the phone) telnet_level: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged ### New Parameters added in Release 4.0 ### # XML URLs services_url: "" ; URL for external Phone Services directory_url: "" ; URL for external Directory location logo_url: "http://192.168.17.11/asterisk-tux.bmp" ; URL for branding logo to be used on phone display # HTTP Proxy Support http_proxy_addr: "" ; Address of HTTP Proxy server http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default) # Dynamic DNS/TFTP Support dyn_dns_addr_1: "" ; restricted to dotted IP dyn_dns_addr_2: "" ; restricted to dotted IP dyn_tftp_addr: "" ; restricted to dotted IP # Remote Party ID remote_party_id: 0 ; 0-Disabled (default), 1-Enabled ### New Parameters added in Release 4.4 ### # Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) call_hold_ringback: 0 ; Default 0 (Disable ringback of held call) argon:/tftpboot# cat SIP00115C407FA3.cnf # SIP Configuration Generic File # Line 1 line1_name: Scott line1_authname: "scott" line1_password: "scott" # Line 2 line2_name: Scott1 line2_authname: "scott1" line2_password: "scott1" # Line 3 line2_name: "Line 2" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 4 line2_name: "Line 4" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 5 line2_name: "Line 5" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 6 line2_name: "Line 6" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" ### New Parameters added in Release 2.0 ### # All user_parameters have been removed # Phone Label (Text desired to be displayed in upper right corner) phone_label: "" ; Has no effect on SIP messaging # Line 1 Display Name (Display name to use for SIP messaging) line1_displayname: "User ID" # Line 2 Display Name (Display name to use for SIP messaging) line2_displayname: "User ID" ### New Parameters added in Release 3.0 ## # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: "cisco" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none messages_uri: "_6101" argon:/tftpboot# Nabeel Jafferali wrote: I had not looked at the phones settings yet, thanks for the suggestion. The setting indicate that there is no configuration on the second line it is listed as "UNPROVISIONED" Go into the phone and program Line 2 Settings directly, without using the SIPMAC.cnf file. If that works, then your .cnf file is wrong. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Multiple lines on Cisco 7960
Someone on the list spotted the problem, there is a typo in my line definitions. Thanks all Scott Henderson wrote: I set this up manually on the phone and it works just fine so config files ... I attached the complete config files so maybe someone can see what I am missing. argon:/tftpboot# cat SIPDefault.cnf # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 ; # Proxy Server proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy5_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy6_address: "192.168.17.13" ; Can be dotted IP or FQDN # Proxy Server Port (default - 5060) proxy1_port: 5060 proxy2_port: 5060 proxy3_port: 5060 proxy4_port: 5060 proxy5_port: 5060 proxy6_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec ### New Parameters added in Release 2.0 ### # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "192.168.17.11" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: April ; Month in which DST starts dst_start_day: "" ; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: "" ; Day of month in which DST stops dst_stop_day_of_week: Sunday ; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2 ; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 ### New Parameters added in Release 2.1 ### # Backup Proxy Support proxy_backup: "" ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) # Emergency Proxy Support proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ### New Parameters added in Release 2.2 ## # NAT/Firewall Traversal nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled # Outbound Proxy Support outbound_proxy: "" ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is 5060 ### New Parameter added in Release 3.0
[Asterisk-Users] Multiple lines on Cisco 7960
I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple sip debug and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend host=dynamic username=scott1 secret=scott1 context=default mailbox=6101 callerid=Scott Henderson 1 macaddress.cnf # Line 1 line1_name: Scott line1_authname: scott line1_password: scott # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 sip debug output from resetting the phone: Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 Expires: 3600 10 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0045611f Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=scott,realm=asterisk,uri=sip:192.168.17.13,response=7b9f392d15161ef76ae35f283e876497,nonce=0045611f,algorithm=md5 Content-Length: 0 Expires: 3600 11 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 Date: Fri, 07 Jan 2005 02:56:25 GMT Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Date: Fri, 07 Jan 2005 02:56:26 GMT CSeq: 102 NOTIFY Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' argon*CLI The result of this configuration is that I always get the first line line_1 but never the second line. From what I can tell the phone never even tries to register the second line. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ Asterisk-Users mailing list
[Asterisk-Users] Message light on 7960 or in this case no message light
I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip chatter is almost none existent and the sip chatter on the first system that works is quite regular. There is a version difference as follows: The system that is working is: Asterisk 1.0.1 built by [EMAIL PROTECTED] on a i686 running Linux The system that isn't working is: Asterisk 1.0.2 built by [EMAIL PROTECTED] on a i686 running Linux I have reviewed everything I can think of but now message lights and the chatter that seems to have the Message information doesn't seem to be occurring on the system that isn't work like it is on the working system. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme room calls quiet for some lines/callers
I just tried out the meetme room feature for the first time and found a few issues with the levels. I had three calls, one on an fxs port (TMD400), 2 on fxo ports, one fxs was a 100P the other was on the TDM400. The phone on the fxs port could hear and everyone could hear that line. The two calls on the fxo could barely hear each other. I did a little fiddling with the rx and tx gain settings in zapata.conf and this impacted the over all levels for all lines but I am not sure I completely understand what is happening here and in what direction the tx and rx are effecting things. Does anyone have some guidance on how I should change these settings or if I should be looking at something else. -- Scott Henderson == Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.337.2860, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Current Call information?
I am wondering how I can inquire on the current call status and the codec used for that call. show channel provides information but am I missing something here, I really want to know which codec has been used for the channel. -- Scott Henderson == Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.337.2860, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
