[Asterisk-Users] Asterisk and Cisco 7960
I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 The FWD line works, the Asterisk line doesn't. Any suggestions or pointers to documentation I might have missed? Thanks, Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 The FWD line works, the Asterisk line doesn't. Any suggestions or pointers to documentation I might have missed? Thanks, Ben Wern Hints to make your life easier and maybe solve your problem: 1) Make the username numeric instead of "cisco". Change all appropriate configs. 2) Do you see the Cisco trying to register with Asterisk? Use "tethereal port 5060" to watch what happens. 3) Just for fun, put "nat=1" in the peer for your Cisco. It won't hurt. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sample configs
Hi, We are just getting started setting up an Asterisk VoIP server. We are very experienced with Linux, networking, tcp/ip, etc. However, some existing sample config files for using Cisco VoIP phones with this server would be VERY helpful. Thanks, Travis Microserv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample configs
Did you do make samples? It has some samples that will help. bkw On Fri, 29 Aug 2003, Travis Johnson wrote: > Hi, > > We are just getting started setting up an Asterisk VoIP server. We are very > experienced with Linux, networking, tcp/ip, etc. However, some existing > sample config files for using Cisco VoIP phones with this server would be > VERY helpful. > > Thanks, > > Travis > Microserv > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample configs
Travis Johnson wrote: Hi, We are just getting started setting up an Asterisk VoIP server. We are very experienced with Linux, networking, tcp/ip, etc. However, some existing sample config files for using Cisco VoIP phones with this server would be VERY helpful. Thanks, Travis Microserv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This is a bare minimum assuming you start with a phone will all default config settings. /etc/asterisk/sip.conf == [1234] callerid="Your Name" <1234> context=yourcontext type=friend secret=yourpassword host=dynamic defaultip=youraddress mailbox=1234 /tftpboot/ (presumably, where your phone config files are) == SIPDefault.cnf == proxy1_address: "asterisk.ip.address" proxy_register: 1 SIPyour:mac:address.cnf === line1_name: 1234 line1_authname: "1234" line1_password: "yourpassword" line1_displayname: "Your Name" ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue timeouts
Hey all, Trying to get a queue setup such that if it times out, the call is directed to a different queue, as follows: [sales] Exten => 450,1,Queue(sales) Exten => 450,2,Queue(reception) The timeout on the sales queue is set to 45 seconds. The call does time out after 45 seconds, but it stays in the sales queue, and the sales queue members are rung. Seems like it never returns from the sales queue. I saw some postings from a few months ago that folks were noting this behavior then, and thought it might be resolved by now. If anyone wants to tell me where to look to try to contribute code to solve the problem, I'm happy to do so. It ought to be pretty trivial. Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Ben Wern wrote: I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> Use something like: [1000] callerid="Ben" <1000> context=sip type=friend secret=1234 host=dynamic defaultip=youraddress mailbox=1000 Optionally: qualify=500 canreinvite=no nat=yes And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) You might want this to be: exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Make sure you didn't set the 'outbound proxy' setting on the phone, that will force everything to the proxy. Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 Set all the 'cisco' entries to '1000' in this case. I have several 7960s working with Asterisk, so I can help you out more if you need it. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QOH (quiet on hold)?
Hi all, I'm not yet an * user, but I'm planning to become one soon. Here is a random question for you * experts: I'm often dialed into a teleconference using an external service (dial toll free number, enter PIN, etc). Sometimes I have an incoming call or want to call someone else and then return to the conference. I can't put the conference on hold because MOH will be injected to the conference (this is extremely rude when it happens because the entire conference is shut down until the holding party comes back!). So I'm stuck. What I want, in addition to MOH, is QOH (quiet on hold). Then I can put the conference on hold, no sound will disturb the other participants, and I can return later. Obviously I can make all holds be quiet (no music), but I would prefer to retain MOH as the basic hold function. The QOH would be an additional "feature". I'm likely to be using Cisco phones if that matters. So, can * do this, and if so, how? Can MOH be selectively enabled/disabled by extension? Are there other ways to solve this problem besides QOH? Thanks for everyone's help. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restricting concurrent SIP calls
Yes :) we made available this patch few weeks ago: http://lists.digium.com/pipermail/asterisk-dev/2003-July/001202.html I hope that It will work with newest chan_sip.c cvs version if not we will update it :))) It's working for us more than 6 months without any problems. Best regards MiniTelecom Team David Harris wrote: Is it possible to restrict the number of concurrent calls made to a SIP peer? Or maybe the number of concurrent calls made to a particular extension. This way I can avoid asterisk trying to make more voice calls to my remote SIP gateway then I have bandwidth to handle. /davidh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample configs
Hello, I think this is the best place for samples :) http://www.loligo.com/asterisk/ http://www.loligo.com/asterisk/current/ Best regards Lubo Travis Johnson wrote: Hi, We are just getting started setting up an Asterisk VoIP server. We are very experienced with Linux, networking, tcp/ip, etc. However, some existing sample config files for using Cisco VoIP phones with this server would be VERY helpful. Thanks, Travis Microserv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet8 DTA310
Has anyone been successful in using the DTA310 as provided by Packet8 to work with asterisk? I have gotten it to register with Asterisk but whenever I try to dial a call all I get is silence, when I dial an invalid extension I get a fast busy signal. When looking at the SIP debug it seems that it is transmitting XML.
RE: [Asterisk-Users] Asterisk and Cisco 7960
> My sip.conf entry for the cisco looks like this: > > [cisco] > type=friend > username=cisco > secret=1234 > host=dynamic > defaultip=[The IP of the 7960] > mailbox= > context=sip > callerid="Ben" <1> Try to remove the defaultip= string. Do you get any errors in the console when it is run in verbose mode? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
Andrew, I removed that entry, still no luck. I also altered the config to use a number (101) as the entry name instead. I get: Got SIP response 404 "Not Found" back from 172.16.1.28 SIP/101-e9a4 is circuit-busy on the console when I try to call it. sip show peers shows the node, with an OK status. Ben At 02:09 AM 8/30/2003 -0400, Andrew Joakimsen wrote: > My sip.conf entry for the cisco looks like this: > > [cisco] > type=friend > username=cisco > secret=1234 > host=dynamic > defaultip=[The IP of the 7960] > mailbox= > context=sip > callerid="Ben" <1> Try to remove the defaultip= string. Do you get any errors in the console when it is run in verbose mode? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew, Thanks for your help! I did have the outgoing proxy set -- since I had FWD set up on line 1. I removed all the FWD stuff, and the outgoing proxy. I altered the entry to have the qualify, canreinvite, and nat lines and also altered the user id to be a number. Now I'm able to call other local extensions, but I can't call into the Cisco. But it's progress! I can also call out to FWD, but audio drops after a few seconds. Don't even want to think about getting FWD calls back into the network. exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GS on ebay...
Brian West wrote: 101's for 68.00 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066051&category=11175 102's for 79.95 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066389&category=11175 You still got those? I *think* it was you selling them. . . Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Ben Wern > Sent: Saturday, August 30, 2003 3:02 AM > To: [EMAIL PROTECTED]; [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 > > Andrew, > > Thanks for your help! No problem, that's what the list is for :) > I did have the outgoing proxy set -- since I had FWD set up on line 1. I > removed all the FWD stuff, and the outgoing proxy. I altered the entry to > have the qualify, canreinvite, and nat lines and also altered the user id > to be a number. Now I'm able to call other local extensions, but I can't > call into the Cisco. But it's progress! > > I can also call out to FWD, but audio drops after a few seconds. Don't > even > want to think about getting FWD calls back into the network. Change your dial strings end in ,Tt) or ,Ttr) > >exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > This didn't work - what does the @1000 indicate? It shouldn't be there, If it's defined as 1000 in sip.conf make your dial string exten => 1000,1,Dial(SIP/1000,20,Ttr) I don't know what the Tt does (lack of documentation) but adding an r has asterisk generate the ringing (when dialing calls to outside providers/cards) and m will insert music on hold in most cases. BEGIN:VCARD VERSION:2.1 N:Joakimsen;Andrew FN:Andrew Joakimsen ([EMAIL PROTECTED]) ORG:Envision Studio TEL;WORK;VOICE:(888) 210-8063 TEL;CELL;VOICE:(305) 776-0334 TEL;WORK;FAX:(305) 669-6720 EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20030819T050904Z END:VCARD
[Asterisk-Users] OT: My congestion music.
Wanna cheap laugh? IAXTel: 17005334094. -Josh
[Asterisk-Users] Installation Problem
Hi, I am quite new to Asterisk and Linux in general. When l try to install the Zaptel component, l get the following error: asterisk:/usr/src/zaptel# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTANDALONE_ZAPATA -c zaptel.c In file included from zaptel.c:36: /usr/include/linux/module.h:21: linux/modversions.h: No such file or directory make: *** [zaptel.o] Error 1 asterisk:/usr/src/zaptel# Can anyone point me in the right direction. Cheers, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installation Problem
Depending on your distribution, you will either need to install your own kernel, or install the kernel source. it should come as a package for your distro. On Sat, Aug 30, 2003 at 08:23:08PM +1000, Phillip Britt wrote: > Hi, > > I am quite new to Asterisk and Linux in general. When l try to install the > Zaptel component, l get the following error: > > asterisk:/usr/src/zaptel# make > cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o > gendigits.o gendigits.c > cc -o gendigits gendigits.o -lm > ./gendigits > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes > -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DECHO_CAN_MARK2 > -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 > -DSTANDALONE_ZAPATA -c zaptel.c > In file included from zaptel.c:36: > /usr/include/linux/module.h:21: linux/modversions.h: No such file or > directory > make: *** [zaptel.o] Error 1 > asterisk:/usr/src/zaptel# > > > Can anyone point me in the right direction. > > Cheers, > Phil > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installation Problem
On Sat, 2003-08-30 at 12:23, Phillip Britt wrote: > Hi, > > I am quite new to Asterisk and Linux in general. When l try to install the > Zaptel component, l get the following error: > > asterisk:/usr/src/zaptel# make > cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o > gendigits.o gendigits.c > cc -o gendigits gendigits.o -lm > ./gendigits > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes > -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DECHO_CAN_MARK2 > -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 > -DSTANDALONE_ZAPATA -c zaptel.c > In file included from zaptel.c:36: > /usr/include/linux/module.h:21: linux/modversions.h: No such file or > directory > make: *** [zaptel.o] Error 1 > asterisk:/usr/src/zaptel# Looks like you have not installed any kernel source to me. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference without zaptel??
Hi, Just need to check somthing.. Am I correct in saying that conferencing does not work on a system that does not have a Digium board installed?? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference without zaptel??
> Am I correct in saying that conferencing does not work on a system that > does not have a Digium board installed?? No, it does. You must download the source for zaptel and around line 90 in Makefile find this string MODULES=.. #ztdummy Uncomment ztdummy, compile and then compile asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference without zaptel??
WipeOut . wrote: Am I correct in saying that conferencing does not work on a system that does not have a Digium board installed?? It doesn't work on a system that doesn't have a zaptel driver installed. If you don't have any zaptel hardware, you can compile a dummy zaptel driver which will let you do conferencing. Download the zaptel module from ftp.asterisk.org (or cvs.digium.org). Edit the Makefile - uncomment the ztdummy.o part in the MODULES line by removing the hash in front of it. make clean && make install the zaptel driver. You might need to make sure you have the appropriate stuff in your kernel, such as the USB UHCI support (not the alternative JE version). depmod -a should give no unresolved symbols. Install the ztdummy module with modprobe ztdummy. You can check it all worked by going lsmod. HTH, -- Alastair Maw <[EMAIL PROTECTED]> MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference without zaptel??
> > > > Am I correct in saying that conferencing does not work on a system > that > > does not have a Digium board installed?? > > No, it does. You must download the source for zaptel and around line 90 > in Makefile find this string > > MODULES=.. > #ztdummy > > > Uncomment ztdummy, compile and then compile asterisk. > I have been unsucessfull with using ztdummy in the past, for some reason when I enabled ztdummy in the past it broke my IAX2 trunks, Although I was told it was a configuration error I could not find any way to fix it so I abandoned ztdummy in favour of having the trunks working.. So I guess I will have to abandon conferencing too then till we are big enough to use an E100P instead of chan_capi.. Thanks anyway.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 DTA310
Post the sip debug .. maybe someone will help you. Martin On Sat, 30 Aug 2003, Andrew Joakimsen wrote: > Has anyone been successful in using the DTA310 as provided by Packet8 to > work with asterisk? I have gotten it to register with Asterisk but > whenever I try to dial a call all I get is silence, when I dial an > invalid extension I get a fast busy signal. When looking at the SIP > debug it seems that it is transmitting XML. > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incomming call issue
I have an issue getting any incomming calls
When the phone rings something picks up and gives it a fast busy.
There is no one using Zap/2
it does the same thing with voicemail and voicemail 2
you can see the console output below,
I would love any help anyone could shead on this issue,
Michael
NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing SetMusicOnHold("Zap/1-1", "default") in new stack
-- Executing Dial("Zap/1-1", "Zap/2|30") in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
-- Executing VoiceMail2("Zap/1-1", "u100") in new stack
-- Playing 'voicemail/default/100/unavail'
WARNING[1192484144]: File chan_zap.c, Line 2853 (zt_handle_event):
Ring/Off-hook in strange state 6 on channel 1
-- Playing 'vm-intro'
-- Playing 'beep'
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: gsm,
0x80ce6a0
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: wav49,
0x80cf1e8
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: wav,
0x80de0d8
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GS on ebay...
Nope wasn't me selling them... I see them on ebay and thought I would pass the info along search ebay for "sip phone" they are still listed. bkw On Sat, 30 Aug 2003, Brian Capouch wrote: > Brian West wrote: > > 101's for 68.00 > > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066051&category=11175 > > > > 102's for 79.95 > > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066389&category=11175 > > > > You still got those? > > I *think* it was you selling them. . . > > Thx. > > B. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Filling PHP Variable from EXTENSION in AGI
Hellooo... Is it possible to fill a variable of PHP-based-AGI-script from dialed extension ? This is what I need to achieve: If someone dial an extension, say 777, I want the dialed extension (777) be filled into PHP variable. I need the dialed extension become a condition of PHP script. Help please... Thanks romsun _ This mail sent using V-webmail - http://www.v-webmail.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incomming call issue
On Sat, 2003-08-30 at 12:38, Lists wrote:
> I have an issue getting any incomming calls
>
> When the phone rings something picks up and gives it a fast busy.
>
> There is no one using Zap/2
>
> it does the same thing with voicemail and voicemail 2
>
> you can see the console output below,
>
> I would love any help anyone could shead on this issue,
>
> Michael
>
> NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2
> (Ring/Answered)...
> -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack
> -- Executing Dial("Zap/1-1", "Zap/2|30") in new stack
> -- Called 2
> -- Zap/2-1 is ringing
> -- Hungup 'Zap/2-1'
> == No one is available to answer at this time
> -- Executing VoiceMail2("Zap/1-1", "u100") in new stack
> -- Playing 'voicemail/default/100/unavail'
> WARNING[1192484144]: File chan_zap.c, Line 2853 (zt_handle_event):
> Ring/Off-hook in strange state 6 on channel 1
> -- Playing 'vm-intro'
> -- Playing 'beep'
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: gsm,
> 0x80ce6a0
> -- x=1, open writing:
> /var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: wav49,
> 0x80cf1e8
> -- x=2, open writing:
> /var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: wav,
> 0x80de0d8
You didn't say what hardware you are using. It sounds a lot like you may
be using a T1 or E1 interface and it is having timing problems and is
going into alarm. When the trunk goes into alarm, your channel bank
would provide a fast busy signal.
--
Steven Critchfield <[EMAIL PROTECTED]>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restricting concurrent SIP calls
On Fri, 2003-08-29 at 23:27, Lubomir Christov wrote: > we made available this patch few weeks ago: > http://lists.digium.com/pipermail/asterisk-dev/2003-July/001202.html Any chance of this making it into the main source? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller Id Issues
Can someone please assist me in getting caller id working with the dev kit. Is there a howto anywhere? I think I have it setup right, it just does not work. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
