Re: [asterisk-users] read write or only read fields in cdr?
Mike Lynchfield a écrit :
try not using dst.. maybe its a regex on te fieldname that matches for
reserved keywords..
try pre_dest instead
On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:
Hello,
I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.
In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
and when I call, all goes fine except that pre_dst has always NULL value
in cdr.
Do you know why?
Is something wrong I did?
I know that original fields in cdr are only readable, but in this cas
pre_dst is one I created myself !!!
Thank you.
I tried yet pre_dst un cdr table.
I even change it to foo or coco or whatever, but nothing is written in
this custom field except NULL.
+-+++--+
| clid| dstchannel | coco | dst
|
+-+++--+
| 2007-02-28 13:31:43 | SIP/x-081b1fd8 | NULL x |
thanks.
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Re: [asterisk-users] read write or only read fields in cdr?
Edgar Luna a écrit :
Hi,
On Wed, 2007-02-28 at 23:43 +0100, Bayrouni wrote:
Hello,
In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
and when I call, all goes fine except that pre_dst has always NULL value
in cdr.
Do you know why?
Is something wrong I did?
As far as I know, custom fields doesn't work with any database backend,
only with CSV. There is an addon in the bug tracker but seems that it
isn't finished.
Regards.
Thank you for this information.
Regards
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[asterisk-users] DTMF not being detected with 1 provider. Works with the other provider...
Hi all! Working on the following brain-scratcher. I am setting up a Trixbox system for someone who uses 'provider A'. Everything works fine, except for the IVR: keypresses by callers are not being detected. Just for testing I added my own provider, 'provider B' to their system. And then the IVR works! Is there any possibility that the config on the provider-side is causing this difference? If yes, what could it be, and is there a way for me to fix this? Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens HiPATH 3700 with Asterisk
Hi, I will like to know if anyone would guide me about how I can to interconnect one SIEMENS HiPATH 3700 with Asterisk. HiPATH have VoIP card and my idea is to do one un IP trunk between them so we would to transfer calls and services (voicemail, IVR,..) between both. We havent PRI ports unused in HiPATH so cheapest method of interconnection is one IP trunk. Any help or comment about will be interesting. Thnks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get values of local channels context
From: Yuan LIU [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 21:24:56 -0800
From: kjcsb [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 18:23:46 -0800 (PST)
Check out /path/to/src/asterisk/doc/README.variables
${DIALEDPEERNUMBER} would give it to me if I sliced it up.
exten = s,n,Set(Foo=${CUT(DIALEDPEERNUMBER,@,2)})
exten = s,n,Set(Foo=${CUT(Foo,/n,1)})
Are there any better options?
Cameron
This is beautiful. How much better can it get? Syntax-wise, you can
combine the two lines into one because the value of CUT() can be used
inside another function including CUT(). Using inheritable
Take this back; CUT expects a variable name, not a value:-(
variable could be an alternative. But efficiency difference is neglegible.
Yuan Liu
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[asterisk-users] transfer function
Hello!
I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
but only calling party can do forward. How to configure '*' to take this
possibility to called party?
ps
both calling/called use sip
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Re: [asterisk-users] about bluetooth channel
On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote: Iban Lopetegi Zinkunegi wrote: 28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the next lines: I thought chan_bluetooth only worked with 1.4 head? You thought wrong, he is talking about chan_bluetooth you are talking about chan_cellphone. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Not registering Port with VSP
**All, ** **I'm guessing no one knows the answer as to why when I register with a **VSP I am not sending a Port number with the registration but only my IP **address. If anyone has any answers it would be greatly appreciated. ** **From: [EMAIL PROTECTED] **[mailto:[EMAIL PROTECTED] On Behalf Of **Klaverstyn, David C **Sent: Wednesday, 28 February 2007 11:08 AM **To: Asterisk Users Mailing List - Non-Commercial Discussion **Subject: [asterisk-users] Not registering Port with VSP ** **Hello All, ** **For some reason my asterisk server is not registering a port number with **my VSPs. This is causing problems where people are not able to dial in **from any of my SIP or IAX VSPs. ** **I do have one VSP that has hard coded my IP and port so I can get **incoming calls but this still leaves a problem with my other VSPs. ** **Hose can I get asterisk to register my IP and port? I have been told **that my asterisk server is registering my IP with the VSP but the port **is empty. Asterisk uses IPv4 addresses when constructing the SIP Contact: header. As a result, it can omit the port declaration if Asterisk is using UDP port 5060. It can do this since an IPv4 address will never resolve to DNS SRV query (thereby bypassing a sticky situation illustrated in RFC 3261 Section 19.1.4 paragraph 9). Your VSP's software should assume UDP port 5060 for your port if you don't explicitly declare it. If your VSP really has such a restrictive SIP Registrar, then you have 2 choices to force a port declaration by Asterisk: 1) Edit channels/chan_sip.c:build_contact() and remove the check to see if the port used by Asterisk is 5060. 2) Change the port Asterisk uses for SIP. This is accomplished by adding a bindport= where is the decimal number of another port besides 5060. Since the new port is not 5060, Asterisk will explicitly declare the new port in the Contact: header. If you choose to use option 2) then you will need to inform the VSP that has your IP/Port hardcoded to update their contact info. Additionally, if you had any Port Forwarding setup on your Router/NAT device, you will need to update that information as well. See ya... d.c. ___ --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] FAX using T38
So does asterisk (Albeit with a commercial package) http://www.attractel.com/t38.html Lee Howard wrote: Matt Riddell [NZ] wrote: Does OpenPBX do a T.38 gateway then? Yes, it does. Lee. ___ --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] 1.4 lost internet internal phones loose registration
Eric ManxPower Wieling wrote: Asterisk gets very upset when DNS is down. You might want to confirm that /etc/hosts has entries for ALL interfaces in that system. That should cause the system to not issue a DNS request to resolve local IPs. Asterisk also seems to barf if it makes a registration/renewal request and it doesn't receive a reply in a timely fashion which will obviously happen if your internet connection disappears. (all versions I've used). ___ --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] FAX using T38
Only Asterisk 1.4.0 has chan_sip.c with T.38 code in it. Although it still doesn't work because it's full of bugs. Seems to me that developers just pasted the T.38 patch code from the branch developing that issue, and nothing else have done to improve it. It has to be debugged. Regards, Ricardo. Zoa wrote: So does asterisk (Albeit with a commercial package) http://www.attractel.com/t38.html Lee Howard wrote: Matt Riddell [NZ] wrote: Does OpenPBX do a T.38 gateway then? Yes, it does. Lee. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: queue information into db
nik600 wrote: actually it isnìt released under any type of licence. if you want i can put the code on my web site (but no earlier than the next week) Please do. And it wouldn't hurt if you, somewhere on the page, put that is released under GPL or something similar. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: fax support
Olle E Johansson wrote: However, the 1.4.0 release is buggy, so either use 1.4 from subversion or wait for 1.4.1. Have you put this information somewhere on web page of Asterisk? I think its fair enough to say - look, this doesn't work as it should, use 1.2X or 1.4 from subversion. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP interface status and calllimit
James Fromm wrote: I've reviewed the bugs reports. I didn't see anything that applied to this. Have you? Could you point it out to me? Just for the record, I believe this is what you are looking for. http://bugs.digium.com/view.php?id=8800 -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with Calling Name ID in SIP: Asterisk sets Caller ID Number as Name if NO Name
Hi, fyi, I use Asterisk 1.2.9.1 In some scenarios, we receive call from PSTN without Callerd ID Name (which is normal). I would like to transfer this call to another softswitch. Again, I would like to let this this CallerID Name Empty. If I look at the logs, I can see -- Executing Macro(SIP/localdomain.com-b79242f0, set-callerid-name) in new stack -- Executing Set(SIP/localdomain.com-b79242f0, CALLERNAME_TMP=) in new stack -- Executing GotoIf(SIP/localdomain.com-b79242f0, 1?format_empty|1) in new stack -- Goto (macro-set-callerid-name,format_empty,1) -- Executing Set(SIP/localdomain.com-b79242f0, CALLERNAME=) in new stack -- Executing Goto(SIP/localdomain.com-b79242f0, set_callername|1) in new stack -- Goto (macro-set-callerid-name,set_callername,1) -- Executing Set(SIP/localdomain.com-b79242f0, CALLERID(name)=) in new stack So, it should be alright! Then I forward the call: -- Executing Dial(SIP/localdomain.com-b79242f0, SIP/990003726831598@next-hop|30) in new stack And if I look into SIP debug mode: Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 11 lines Reliably Transmitting (no NAT) to XXX:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDPXXX:5060;branch=z9hG4bK6dfcce4c;rport From: *0037253415630* sip:[EMAIL PROTECTED];tag=as4479803d To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: GSMDuo-VM Max-Forwards: 70 Date: Thu, 01 Mar 2007 11:05:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY P-Asserted-Identity: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 236 As you can see, name is set ! eventhough callerid(name)= Is it a bug ? How can I really clear this callerid(name) ? How to prevent Asterisk to put back as Name, the number ? Thanks for your kind return ! Regards, Jean -Marc -- Called [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400p Loaded only once
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
This is part of my dmesg
audit(1172747900.510:5): avc: denied { net_bind_service } for pid=1657
comm=hidd capability=10 scontext=system_u:system_r:bluetooth_t:s0
tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
eth0: no IPv6 routers present
[drm] Initialized drm 1.1.0 20060810
ACPI: PCI Interrupt :01:00.0[A] - GSI 16 (level, low) - IRQ 19
[drm] Initialized r128 2.5.0 20030725 on minor 0
agpgart: Found an AGP 3.0 compliant device at :00:00.0.
agpgart: Device is in legacy mode, falling back to 2.x
agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
audit(1172747921.184:6): avc: denied { getattr } for pid=2323
comm=pam_console_app name=card0 dev=tmpfs ino=7969
scontext=system_u:system_r:pam_console_t:s0-s0:c0.c255tcontext=system_u:object_r:device_t:s0
tclass=chr_file
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.14
Zaptel Echo Canceller: KB1
and finally this is my configuration
fxsks=1-4
loadzone=us
defaultzone=us
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[asterisk-users] Revolution Call Accounting Desktop
Has anyone used this billing application with Asterisk? I have one potential costumer (hotel) that will use application that connects with Fidelio/Micros and so they can use Revolution Call Accounting Desktop for billing. More info about product you can find on this page http://www.telecost.com/revcall.htm Now, has anybody implemented this with Asterisk? Any known issues? Is it ready for production system? I would appreciated any info about this. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail advanced options problem with mysql datbase
Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001 = ,, [test2] 2001 = yyy,,, Error Message showing: No mailbox number '2001' in context 'test1', no reply sent The above problem occuring when i was reading my mailbox and when i try to send a reply to the person who sent me the message using advanced options no1 Can anybody plaease help me out? Thanks in advace Srinivas Antarvedi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UK SIP Gateway
Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. Apologies if this is the incorrect forum for this type of request. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] TDM400p Loaded only once
On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote:
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
This is what you attempt to configure
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
And that is the result.
What do you see on /proc/zaptel ?
What is the output of 'xpp/utils/genzaptelconf -l' from the zaptel
source directory?
This is part of my dmesg
audit(1172747900.510:5): avc: denied { net_bind_service } for pid=1657
comm=hidd capability=10 scontext=system_u:system_r:bluetooth_t:s0
tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
eth0: no IPv6 routers present
[drm] Initialized drm 1.1.0 20060810
ACPI: PCI Interrupt :01:00.0[A] - GSI 16 (level, low) - IRQ 19
[drm] Initialized r128 2.5.0 20030725 on minor 0
agpgart: Found an AGP 3.0 compliant device at :00:00.0.
agpgart: Device is in legacy mode, falling back to 2.x
agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
audit(1172747921.184:6): avc: denied { getattr } for pid=2323
comm=pam_console_app name=card0 dev=tmpfs ino=7969
scontext=system_u:system_r:pam_console_t:s0-s0:c0.c255tcontext=system_u:object_r:device_t:s0
tclass=chr_file
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.14
Zaptel Echo Canceller: KB1
Load of zaptel. No load of wctdm in sight.
and finally this is my configuration
fxsks=1-4
loadzone=us
defaultzone=us
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] UK SIP Gateway
I have used www.voiptalk.org for a number of years with their IAX2 connectivity and they seem very reliable with no echo issues. They will also change the CID to your number if you fax them proof of ownership. Chris - Original Message - From: --[ UxBoD ]-- [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 01, 2007 11:56 AM Subject: [asterisk-users] UK SIP Gateway Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. Apologies if this is the incorrect forum for this type of request. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3 way calling independent of phone hw.
I'm looking for a recipe for a 3 way call where one of the parties can (without using the flash button) dial-out and add a third participant to the call. I tried Googling but it seems I'm missing a key search term. The reason I wanted to avoid using the flash button is that some handsets don't have it (nokia E61 who's 2 way calling via sip is also broken) Something like: 1. party 1 calls party 2 2. either party 1 or 2 hits * on keypad 3. asterisk prompts for party 3's telephone number 4. asterisk dials party 3. 5. party 3 answers and is immediately added to 3-way call 6. the inviter has the option of pushing # to terminate party 3 (should the call only reach party 3's voicemail). Either that or a ways to do DISA from within the meet-me functionality. I can't imagine I'm the only person with this sort of requirement. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --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] UK SIP Gateway
On Thu, Mar 01, 2007 at 12:17:49PM -, Chris Stenton wrote: I have used www.voiptalk.org for a number of years with their IAX2 connectivity and they seem very reliable with no echo issues. They will also change the CID to your number if you fax them proof of ownership. There's several VoIP players in the UK 1899 Gradwell.com to name a few. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] UK SIP Gateway
I have used www.voiptalk.org for a number of years with their IAX2 connectivity and they seem very reliable with no echo issues. They will also Second that. Not cheap but reliable and been there for years. ___ --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] Help Needed: Can't make local calls on a brand new PRI
Yes asterisk was stopped and restarted. ztcfg was not rerun. I've never had to rerun that when I made changes in that file before, but we can try it. MY WISH: TelCo Switchmen could talk intelligently about the protocols used on PRIs! On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: It is currently set to unknown. switchtype=national signalling=pri_cpe pridialplan=unknown Was it originally or did you just change it? Did you stop Asterisk and do ztcfg after making the changes? Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cannot hear ringback music from telco
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to
the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music, users
reported that they could not hear the music but can only hear the standard
ring tone generated by the system.
Is there any kind of settings need to allow the ringback music pass to the
phone? We only use this simple dialplan to make outgoing call:
exten = _9.,1,Set(CALLERID(all)=${CIDPREFIX}${CALLERIDNUM})
exten = _9.,n,Dial(Zap/g0/${EXTEN:1}||T)
* CIDPREFIX is the 5 digit prefix of the company's PRI line, CALLERIDNUM
are 3 digit internal numbers
Thanks!
Vincent
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Re: [asterisk-users] about bluetooth channel
Dave Cotton wrote: On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote: Iban Lopetegi Zinkunegi wrote: 28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the next lines: I thought chan_bluetooth only worked with 1.4 head? You thought wrong, he is talking about chan_bluetooth you are talking about chan_cellphone. Yeah, I realized that after I posted. I apologize if I confused anyone more than myself. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot hear ringback music from telco
Vincent Tam wrote:
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect
to the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music,
users reported that they could not hear the music but can only hear
the standard ring tone generated by the system.
Is there any kind of settings need to allow the ringback music pass to
the phone? We only use this simple dialplan to make outgoing call:
exten = _9.,1,Set(CALLERID(all)=${CIDPREFIX}${CALLERIDNUM})
exten = _9.,n,Dial(Zap/g0/${EXTEN:1}||T)
* CIDPREFIX is the 5 digit prefix of the company's PRI line,
CALLERIDNUM are 3 digit internal numbers
Thanks!
Vincent
Use the m option in your dial statement.
*m*: Provide Music on Hold to the calling party until the called channel
answers. This is mutually exclusive with option 'r', obviously. Use
m(class) to specify a class for the music on hold.
Thanks,
Steve Totaro
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Re: [asterisk-users] Cannot hear ringback music from telco
Vincent Tam wrote:
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect
to the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music,
users reported that they could not hear the music but can only hear
the standard ring tone generated by the system.
Is there any kind of settings need to allow the ringback music pass to
the phone? We only use this simple dialplan to make outgoing call:
exten = _9.,1,Set(CALLERID(all)=${CIDPREFIX}${CALLERIDNUM})
exten = _9.,n,Dial(Zap/g0/${EXTEN:1}||T)
Try adding an Answer between those two, just to see who is producing the
ringing. If you do NOT hear ringing but DO hear music, it's an asterisk
or snom problem. If you still hear ringing, it's your PRI.
Might also want to check in to the phone's manual to see if there's a
setting to enable early-audio.
Trevor
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[asterisk-users] IAX best practices
Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)? Regards, Alex ___ --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] SIP to IAX - forcing codec pass thru
Peter Gradwell wrote: mmm, but as you've seen, some customers like using multiple codecs. The cisco kit is able to support a raft of options - and it does transcoding very nicely - so the optimum solution is to have the cisco + customer's asterisk agree on the same codec, and then have our asterisk server (in the middle) do as little as possible. As an example, I'm one such customer who likes to send voice calls to them as G.729 and fax calls as G.711a. IIRC a standard has been defined for end-to-end codec renegotiation, but this hasn't been implemented into asterisk (opr probably anything else) yet. ___ --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] Zaptel 1.4.0
I believe I noticed that I had upgraded the kernel, but not yet restarted. I restarted, and I think that was all I had to do to get it running again. --Mike Message: 14 Date: Wed, 21 Feb 2007 18:01:22 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.0 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Feb 21, 2007 at 09:52:32AM -0600, Mike Hammett wrote: I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don't see any errors. This is out of my modprobe.conf: [ snip ] However: [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Module zaptel not found. Any chance that this is just a missing depmod run? depmod modinfo zaptel Or maybe you installed the modules to an incorrect directory: uname -r find /lib/modules -name zaptel.ko If so, it probably means you built it with incorrect kernel source / configuration. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --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] IAX best practices
Asterisk wrote: Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)? Regards, Alex Bandwidth and latency. IAX2 is remarakably good at traversing NAT and even double NATs. It should just work. The issues that I ran into are low bandwidth and latency. Not much you can do about latency besides getting a better route and putting QoS on your equipment and hoping that your provider either observes your tagging or is not very latent to begin with. The other is bandwidth which I found SPEEX works wonders (but adds to latency). In my experience, bandwidth issues result in choppy audio and latency results in delays which cause people to talk on top of each other and can be extremely annoying. Try pinging a router or device at the remote side to get an idea of how latent your connection will be. Thanks, Steve ___ --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] Zaptel 1.4.0
Mike, Did you tried with make all instead of make linux26? That worked for me on FC5. On FC6, I have to reinstall everything and worked with make all. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Thursday, March 01, 2007 10:00 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Zaptel 1.4.0 I believe I noticed that I had upgraded the kernel, but not yet restarted. I restarted, and I think that was all I had to do to get it running again. --Mike Message: 14 Date: Wed, 21 Feb 2007 18:01:22 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.0 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Feb 21, 2007 at 09:52:32AM -0600, Mike Hammett wrote: I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don't see any errors. This is out of my modprobe.conf: [ snip ] However: [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Module zaptel not found. Any chance that this is just a missing depmod run? depmod modinfo zaptel Or maybe you installed the modules to an incorrect directory: uname -r find /lib/modules -name zaptel.ko If so, it probably means you built it with incorrect kernel source / configuration. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --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] Paid support offered
On Thu, 01 Mar 2007 17:29:57 +1300 Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike Lynchfield wrote: We have decided to allow our tech's to do support for non-clients of voicemeup.com This should normally be kept on the Asterisk-Biz list This list is for Non-Commercial Discussion Yes and no. If he want to advertise on the list, he can actually answer people's questions and put his ads as a footer. Then his ads will be welcome as part of an answer. -- Thanks http://www.sqlhacks.com The SQL knowledge base ___ --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] Snom 320 password
It must be the challenge response bug. They are still using 1.0.x and I turned off challenge response on the phone. I made the change last week, but I haven't heard from the user one way or the other. This server is slated to be upgraded to 1.4.0. --Mike -- Message: 14 Date: Wed, 21 Feb 2007 16:52:10 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [asterisk-users] Snom 320 password To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii A client of mine has a Snom 320. Usually when he comes in each morning, it is asking him for a password. A power cycle brings it back to normal operation. How do I troubleshoot this further? --Mike -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070221/0e4d3a b8/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] blieve i my TE110P or My teleco provider ??
hi eveybody, after many test with your help and the irc channels help, i get the led on TE110P green with this config: span=1,1,0,ccs,ami = alarms OK Green Led but the provider say that i have to set my span to this span=1,1,0,ccs,hdb3,crc4 = alarms: YEL/RED i can't make call's yet to test because they have to sync the Modulator in the other side so any remark? is my card TE110P get crazy? is the TELECO are crazy? any idea ___ --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] Re: queue information into db
On 3/1/07, Tomislav Parcina [EMAIL PROTECTED] wrote: nik600 wrote: actually it isnìt released under any type of licence. if you want i can put the code on my web site (but no earlier than the next week) Please do. And it wouldn't hurt if you, somewhere on the page, put that is released under GPL or something similar. i'm rewriting it in english. probably i'll release it under GPL on sourceforge this night bye ___ --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] Zaptel 1.4.0
Try this: /etc/init.d/zaptel start Than do lsmod |grep zaptel and it should show zaptel loaded Ricardo. Mike Hammett wrote: I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don’t see any errors. This is out of my modprobe.conf: install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg install pciradio /sbin/modprobe --ignore-install pciradio /sbin/ztcfg install ztd-loc /sbin/modprobe --ignore-install ztd-loc /sbin/ztcfg install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg alias wcfxs wctdm alias wct2xxp wct4xxp install zttranscode /sbin/modprobe --ignore-install zttranscode /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg However: [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Module zaptel not found. /var/log/dmesg doesn’t say anything about zaptel. ___ --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] UK SIP Gateway
On Thu, 1 Mar 2007, --[ UxBoD ]-- wrote: Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. There are dozens in the UK. Sipgate is a german company and his termination server (sipgate.co.uk) is actually in Germany, but he uses a UK wholesaler for IP-PSTN connectivity. (I don't know if he then connects back to them via the 'net or if they have a local connection in Germany) I've used them for about a year now and so-far so good, but I've not used it in anger as it were - just for some testing. I also have an account with www.voiptalk.org - same though, just for the occasional test. If you want to call me on my sipgate number, drop me a private email, or you could try http://www.wirelessforums.org/uk-telecom-voip/speaking-clock-1025.html Gradwell is probably the benchmark for these services right now, but they're not cheap, and thats usually what people are looking for IME... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Testing asterisk with sipp
Hi all, I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our asterisk installation. We have a very simple dialplan that uses FastAgi. I'm finding that all calls to GET VARIABLE from the FastAgi are returning null when the dialplan is invoked from sipp -- and they work fine when invoked from a softphone on the same machine, for example. Does anyone have any insight as to what might be going on here? Thanks John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extensions +International
This should be easy, but I can't find the right wildcard.
Right now I have
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
for international and for local
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:1},,wW)
The problem is if the call isn't typed in, then you press dial, we have
problems... Example:
I pick up the handset and get a dialtone. I press 9011331234567 or
something international. Before I can finish, the local option kicks in
because it saw 9.
Is there a way to say
_9[2-9]NXXX or something like that?
Rob
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RE: [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Wednesday, February 28, 2007 21:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk I have a TELUS PRI for a while, resold via Bell... dropped it after a few months due to broken promises and failure to deliver /any/ of the things we said we required when ordering. During this time, I learned that with a TELUS PRI you cannot send name. It's simply dropped at the switch. If you want a name to show up, you have to beg and plead for them to manually update a database of theirs. On top of this, that name will show up on *all* DIDs associated with your block, so you will have to either have no name, a generic name of sorts, or buy individual phone numbers at a higher rate and beg them each time to manually do the update. That doesn't sound promising; however my problem (as shown in the debug output in my original post) is that it appears that the Zaptel drivers aren't even attempting to SEND the name out the PRI, only the number. Is there a setting in the Zapata.conf or zaptel.conf that is necessary to tell it I want it to send the name too? Andrew If you find another way, please do share... but that is 6 months headache we had just to find out that it was impossible to send the name, despite the majority of their technicians stating otherwise. This is in BC. Maybe it's different in Alta? Webster, Andrew wrote: Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't support the name feature, so they've changed my link type). Version: Asterisk 1.2.14 svn rev 48468 Asterisk Log: Executing Set(SIP/304-091aafb8, CALLERID(all)=Andrewnn) in new stack (I've replaced the digits with n). PRI debug shows: Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 21 80 nn nn nn nn nn nn nn nn nn nn] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'nn' ] From zapata.conf: callerid=asreceived ;Sangoma A101 port 1 [slot:12 bus:0 span: 1] switchtype=ni1 context=from-zaptel overlapdial=yes facilityenable=yes group=0 signalling=pri_cpe channel = 1-23 Thanks! -- Andrew ___ --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 ___ --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] Call connected, cannot hear or speak - $20 for fix
Not yet I'll pay for you help. I have been to the sineapps page many times.I know you guys got skillz On 2/28/07, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Supa wrote: I using my provdier like so SIP/Telasip-gw4/5198843344 when bridging calls. All my local extensions work, so does disa and the like Did you get this going? - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5ljXS6d5vy0jeVcRAtiuAJ9m5LOTjFDiPdm+Ux3Ic6nXAPRcaACcDHjC J5Gdt8Rc/BDfi33U8Bku85A= =A2KZ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple simultaneous calls
Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance, Stefano :. CONFIDENZIALE: Questo messaggio e gli eventuali allegati sono confidenziali e riservati. Se vi è stato recapitato per errore e non siete fra i destinatari elencati, siete pregati di darne immediatamente avviso al mittente. Le informazioni contenute non devono essere mostrate ad altri, né utilizzate, memorizzate o copiate in qualsiasi forma. CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium.___ --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] Extensions +International
I pick up the handset and get a dialtone. I press 9011331234567 or something international. Before I can finish, the local option kicks in because it saw 9. Is there a way to say _9[2-9]NXXX or something like that? Are you sure the handset is not processing that call string, making it local before it even gets submitted to Asterisk? I know in the Snom's that that can happen if your number matching is hokey... Stefano ___ --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] UK SIP Gateway
I've been using Voiptalk.org for about a year now and it passes the wife test no problem at all. IAX2 is supported with trunking to save a bit of bandwidth. I use sipgate for an incomming ringback number very very handy. I use Gradwell for domain reg etc and they are excellent I've no reason to think that their VOIP offering is any different. Harvey - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 01, 2007 4:23 PM Subject: Re: [asterisk-users] UK SIP Gateway On Thu, 1 Mar 2007, --[ UxBoD ]-- wrote: Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. There are dozens in the UK. Sipgate is a german company and his termination server (sipgate.co.uk) is actually in Germany, but he uses a UK wholesaler for IP-PSTN connectivity. (I don't know if he then connects back to them via the 'net or if they have a local connection in Germany) I've used them for about a year now and so-far so good, but I've not used it in anger as it were - just for some testing. I also have an account with www.voiptalk.org - same though, just for the occasional test. If you want to call me on my sipgate number, drop me a private email, or you could try http://www.wirelessforums.org/uk-telecom-voip/speaking-clock-1025.html Gradwell is probably the benchmark for these services right now, but they're not cheap, and thats usually what people are looking for IME... Gordon ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About queues and multiple lines.
Hi for all I have one queue and one agent. That agent has a SPA941 with 3 lines configured to an asterisk. That agent logins into queue. If two clients call into that queue and the agent receive the two calls (one for any different line). It is possible that exist any configuration on asterisk to avoid that problem without limiting the number of lines on the agent phone? I'm using trixbox with asterisk 1.2.15. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime
Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. ___ --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] Multiple simultaneous calls
I wouldn't do that with softphones, unless the softphones are designed to do this. The delay will vary depending on the audio card, OS, and drivers. (And the phones might not all answer at the same time, but if you use music on hold or so to play that should not be a problem). [EMAIL PROTECTED] wrote: Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance, Stefano :. CONFIDENZIALE: Questo messaggio e gli eventuali allegati sono confidenziali e riservati. Se vi è stato recapitato per errore e non siete fra i destinatari elencati, siete pregati di darne immediatamente avviso al mittente. Le informazioni contenute non devono essere mostrate ad altri, né utilizzate, memorizzate o copiate in qualsiasi forma. CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Problem with TE212P
-- Forwarded message -- From: Benito Camelas [EMAIL PROTECTED] Date: Wed, 28 Feb 2007 11:21:52 +0100 Subject: Problem with TE212P To: asterisk-users@lists.digium.com Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear 31 TE2/0/1/7 Clear 32 TE2/0/1/8 Clear 33 TE2/0/1/9 Clear 34 TE2/0/1/10 Clear 35 TE2/0/1/11 Clear 36 TE2/0/1/12 Clear 37 TE2/0/1/13 Clear 38 TE2/0/1/14 Clear 39 TE2/0/1/15 Clear 40 TE2/0/1/16 HDLCFCS 41 TE2/0/1/17 Clear 42 TE2/0/1/18 Clear 43 TE2/0/1/19 Clear 44 TE2/0/1/20 Clear 45 TE2/0/1/21 Clear 46 TE2/0/1/22 Clear 47 TE2/0/1/23 Clear 48 TE2/0/1/24 Clear 49 TE2/0/1/25 Clear 50 TE2/0/1/26 Clear 51 TE2/0/1/27 Clear 52 TE2/0/1/28 Clear 53 TE2/0/1/29 Clear 54 TE2/0/1/30 Clear 55 TE2/0/1/31 Clear cat /proc/zaptel/3 Span 3: TE2/0/2 T2XXP (PCI) Card 0 Span 2 56 TE2/0/2/1 Clear 57 TE2/0/2/2 Clear 58 TE2/0/2/3 Clear 59 TE2/0/2/4 Clear 60 TE2/0/2/5 Clear 61 TE2/0/2/6 Clear 62 TE2/0/2/7 Clear 63 TE2/0/2/8 Clear 64 TE2/0/2/9 Clear 65 TE2/0/2/10 Clear 66 TE2/0/2/11 Clear 67 TE2/0/2/12 Clear 68 TE2/0/2/13 Clear 69 TE2/0/2/14 Clear 70 TE2/0/2/15 Clear 71 TE2/0/2/16 HDLCFCS 72 TE2/0/2/17 Clear 73 TE2/0/2/18 Clear 74 TE2/0/2/19 Clear 75 TE2/0/2/20 Clear 76 TE2/0/2/21 Clear 77 TE2/0/2/22 Clear 78 TE2/0/2/23 Clear 79 TE2/0/2/24 Clear 80 TE2/0/2/25 Clear 81 TE2/0/2/26 Clear 82 TE2/0/2/27 Clear 83 TE2/0/2/28 Clear 84 TE2/0/2/29 Clear 85 TE2/0/2/30 Clear 86 TE2/0/2/31 Clear Before I do load the modules the leds are ligthing. But after a ztcfg -v the led of the second span is off. First I do insmod wct4xxp and after ztcfg -vv. The zaptel.conf file is like this: # #zaptel.conf # fxsks=1-24 span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=25-39,41-55 dchan=40 bchan=56-70,72-86 dchan=71 loadzone=nl defaultzone=nl (I have an TDM24P too, it works ok). In this moment the led of the first span of the TE212P is in RED (if no cable connected) or in GREEN (if a cable is conected), but the led of the second span is off. This is shown in a pri show span in the CLI (with no cable connected): pri show span 2 Primary D-channel: 40 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 *CLI pri show span 3 Primary D-channel: 71 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 If anybody knows what'ś the problem I'll be very pleasent for your help. Best Regards, Benito ___ --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] Extensions +International
Yeh, I'm sure. If I watch the debug logs in *, I see each digit running checks to see if it matches a dialplan yet. :) Rob McGhee, Stefano wrote: I pick up the handset and get a dialtone. I press 9011331234567 or something international. Before I can finish, the local option kicks in because it saw 9. Is there a way to say _9[2-9]NXXX or something like that? Are you sure the handset is not processing that call string, making it local before it even gets submitted to Asterisk? I know in the Snom's that that can happen if your number matching is hokey... Stefano ___ --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] Multiple simultaneous calls
[EMAIL PROTECTED] wrote: Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance, Stefano I suppose you could do that although, I am unclear on the auto-answering softphone and the loudspeaker thing. Is this just for overhead paging or something? You could put all the phones in a ring group with ringall and use the computer's sound card to connect to an amplified speaker setup. You could also look at ices2 to stream audio or some other streaming technology. Thanks, Steve ___ --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] Problem with TE212P
Hi, if no cable is connected you'll get the red alarm can you tell me the schema of your corssover cable. try yo connect the 2 slots berween them and make internal calls(i 'm not sure of this option but somemone called shimi has asked me if i have 2 TE110P to test if my cable my config are ok) i hope this will be helpful, i still a newbie :) ___ --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] about bluetooth channel
Iban, For me, it seems to be the codec. Which one are you using? On 3/1/07, Steve Totaro [EMAIL PROTECTED] wrote: Dave Cotton wrote: On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote: Iban Lopetegi Zinkunegi wrote: 28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the next lines: I thought chan_bluetooth only worked with 1.4 head? You thought wrong, he is talking about chan_bluetooth you are talking about chan_cellphone. Yeah, I realized that after I posted. I apologize if I confused anyone more than myself. Thanks, Steve ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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] Asterisk Realtime
Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I can’t see any signs that it’s working. I followed and double-checked a few different guides around the net, but haven’t been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Help understanding SIP SHOW CHANNELS
All the calls you listed in your example were simultaneous calls *from the same user*. Is this what you were intending? I'm assuming this 'cause the Peer column contains the same IP address each time, and the User column contains the same User number. Only the Call ID changes. Moreover, in the Message column, each channel is in INVITE state. I don't expect that that's typical. Moj Michelle Dupuis wrote: I have a high volume asterisk 1.40 installation and I ran a SIP SHOW CHANNELS. (see partial output below). My questions are: 1. wc-l of the output shows 4000 lines. Does this mean 2000 active calls? (2 channels per call) 2. The latter part of the output shows unkn for Form column. Why does it not know the codec? Could it be UDPTL? Or are these calls messed up? 3. I see a lot of WARNING[20224]: udptl.c:819 ast_udptl_new_with_bindaddr: No UDPTL ports remaining errors - is this related to number 2 above? Thanks, MD Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 172.16.116.29 2897516#15 1db77942648 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#13 585240b13ef 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#13 0244b8e668d 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#16 46d0960e602 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#12 66a6c012658 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#15 53e3fc8e2e9 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#18 5fb1a13f19a 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 367737ca0a5 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#18 5af77b3d2a2 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#14 0e85b13b166 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#18 57a534fe7e3 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#14 7b9c71a27d5 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#19 07e8c69614a 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 2695982906d 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#15 17e662d330d 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#12 0594e5f75b9 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#19 3f50957f643 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#18 2a7f856b76d 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#19 13661f065e3 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 6120b54c57c 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#19 6784f95a35a 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 27a57bf82f0 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#19 6660378d7a1 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#14 2441304239a 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 6956c7fb2e3 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#15 40f7f2f6653 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 65670c8c652 00102/0 unkn No Init: INVITE 172.16.116.29 2897516#15 3afc8920231 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#14 2332d1bc257 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#19 7fbb17ff3b1 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#14 761eaa923ab 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 1a1413b75b0 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#14 2416bac8174 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#16 2054fa890a1 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#15 099778402f9 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 4a2c127f14e 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#17 07aa1be846b 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#18 0036884c158 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#15 56cd1cba00b 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#16 04c9881555c 00102/0 g729 No Init: INVITE 172.16.116.29 2897516#18 4a91a496569 00102/0 unkn No Init: INVITE 172.16.116.29
[asterisk-users] Tesco Internet Phone
I've gotten hold of a Tesco Internet Phone which is a dect phone with the base connecting to the pc via usb. Has anyone been able to get this working with any softphone like xlite ? It seems as if the tesco internet phone uses IAX - the software that comes with it is a rebranded firefly (or so it seems) I already have a SPA2000 and SPA3000 hooked up, but I was just curious to see if I could get this to work. Julian. ___ --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] about bluetooth channel
What do you mean by codec? i am using the release posted by Theo in http://crazygreek.co.uk/content/chan_bluetooth. From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] about bluetooth channel Date: Thu, 1 Mar 2007 14:55:26 -0300 Iban, For me, it seems to be the codec. Which one are you using? On 3/1/07, Steve Totaro [EMAIL PROTECTED] wrote: Dave Cotton wrote: On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote: Iban Lopetegi Zinkunegi wrote: 28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the next lines: I thought chan_bluetooth only worked with 1.4 head? You thought wrong, he is talking about chan_bluetooth you are talking about chan_cellphone. Yeah, I realized that after I posted. I apologize if I confused anyone more than myself. Thanks, Steve ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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 _ Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349 ___ --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] Tesco Internet Phone
The phones are provide by Freshtel in Australia if that's any help. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, 1 March 2007 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Tesco Internet Phone I've gotten hold of a Tesco Internet Phone which is a dect phone with the base connecting to the pc via usb. Has anyone been able to get this working with any softphone like xlite ? It seems as if the tesco internet phone uses IAX - the software that comes with it is a rebranded firefly (or so it seems) I already have a SPA2000 and SPA3000 hooked up, but I was just curious to see if I could get this to work. Julian. ___ --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] Re: queue information into db
i'm sorry but due to some problem the software will be released not first than Wednesday 7/02/2007. i'll post a message . bye On 3/1/07, nik600 [EMAIL PROTECTED] wrote: On 3/1/07, Tomislav Parcina [EMAIL PROTECTED] wrote: nik600 wrote: actually it isnìt released under any type of licence. if you want i can put the code on my web site (but no earlier than the next week) Please do. And it wouldn't hurt if you, somewhere on the page, put that is released under GPL or something similar. i'm rewriting it in english. probably i'll release it under GPL on sourceforge this night bye ___ --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] Tesco Internet Phone
Yeah, that's where firefly comes from, doesn't it. I've got the base station plugged in, and the handset connected to it, but it always says pc unavailable. My system (xp) sees a usb phone for speakers and microphone, but I can't get it to work. Julian. Dean Collins wrote: The phones are provide by Freshtel in Australia if that's any help. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, 1 March 2007 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Tesco Internet Phone I've gotten hold of a Tesco Internet Phone which is a dect phone with the base connecting to the pc via usb. Has anyone been able to get this working with any softphone like xlite ? It seems as if the tesco internet phone uses IAX - the software that comes with it is a rebranded firefly (or so it seems) I already have a SPA2000 and SPA3000 hooked up, but I was just curious to see if I could get this to work. Julian. ___ --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 ___ --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] Asterisk Realtime
Brian Capouch wrote: Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. Or put console = notice,warning,error,verbose,debug in logger.conf / run asterisk -vvvdddc :) This will give you all MySQL queries and warnings. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Tesco Internet Phone
plug it in a linux box and tell us what it is please, generic-usb-audio or what? Bails Julian Lyndon-Smith wrote: Yeah, that's where firefly comes from, doesn't it. I've got the base station plugged in, and the handset connected to it, but it always says pc unavailable. My system (xp) sees a usb phone for speakers and microphone, but I can't get it to work. Julian. Dean Collins wrote: The phones are provide by Freshtel in Australia if that's any help. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, 1 March 2007 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Tesco Internet Phone I've gotten hold of a Tesco Internet Phone which is a dect phone with the base connecting to the pc via usb. Has anyone been able to get this working with any softphone like xlite ? It seems as if the tesco internet phone uses IAX - the software that comes with it is a rebranded firefly (or so it seems) I already have a SPA2000 and SPA3000 hooked up, but I was just curious to see if I could get this to work. Julian. ___ --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 ___ --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] Asterisk Realtime
queue show Show status of a specified queue realtime load Used to print out RealTime variables. realtime update Used to update RealTime variables. restart gracefully Restart Asterisk gracefully Aiur*CLI realtime load You must supply a family name, a column to match on, and a value to match to. I am using Asterisk 1.4.0 and MySQL. It appears that the only realtime options are for loading and updating specific items from the database. The only database options seem to be for dundi. Under modules, all I could find is: Aiur*CLI module show like pbx_realtime.so Module Description Use Count pbx_realtime.soRealtime Switch 0 1 modules loaded --Mike -- Message: 12 Date: Thu, 01 Mar 2007 13:02:23 -0500 From: Brian Capouch [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Realtime To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252; format=flowed Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I cant see any signs that its working. I followed and double-checked a few different guides around the net, but havent been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- ___ --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] Tesco Internet Phone
Disregard lat post, suddenly saw DECT, what is the output of lsusb though? Julian Lyndon-Smith wrote: Yeah, that's where firefly comes from, doesn't it. I've got the base station plugged in, and the handset connected to it, but it always says pc unavailable. My system (xp) sees a usb phone for speakers and microphone, but I can't get it to work. Julian. Dean Collins wrote: The phones are provide by Freshtel in Australia if that's any help. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, 1 March 2007 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Tesco Internet Phone I've gotten hold of a Tesco Internet Phone which is a dect phone with the base connecting to the pc via usb. Has anyone been able to get this working with any softphone like xlite ? It seems as if the tesco internet phone uses IAX - the software that comes with it is a rebranded firefly (or so it seems) I already have a SPA2000 and SPA3000 hooked up, but I was just curious to see if I could get this to work. Julian. ___ --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 ___ --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] Asterisk Realtime
What do you have setup in the res_mysql.conf file and extconfig.conf files? Have you installed the asterisk addons for 1.4 to get support for mysql? On 3/1/07, Mike Hammett [EMAIL PROTECTED] wrote: queue show Show status of a specified queue realtime load Used to print out RealTime variables. realtime update Used to update RealTime variables. restart gracefully Restart Asterisk gracefully Aiur*CLI realtime load You must supply a family name, a column to match on, and a value to match to. I am using Asterisk 1.4.0 and MySQL. It appears that the only realtime options are for loading and updating specific items from the database. The only database options seem to be for dundi. Under modules, all I could find is: Aiur*CLI module show like pbx_realtime.so Module Description Use Count pbx_realtime.soRealtime Switch 0 1 modules loaded --Mike -- Message: 12 Date: Thu, 01 Mar 2007 13:02:23 -0500 From: Brian Capouch [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Realtime To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252; format=flowed Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I cant see any signs that its working. I followed and double-checked a few different guides around the net, but havent been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- ___ --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 -- Bruce Nortex Networks ___ --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] Voice mail is not giving unavailable or busy prompts
C F wrote: If the temp message exists then that will play. The user has to log into the mailbox (app_voicemailmain) and select 0 for mailbox options, and delete the temp message. Or you could do it using the shell. I finally resolved this problem by putting format=wav in the general section of the voicemail.conf file. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.1
Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.1
On Thu, 2007-03-01 at 16:39 -0500, Forrest Beck wrote: Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... I understand perfectly what you meant, but just a reminder that asterisk is now under svn, not cvs. Kevin Fleming is in the middle of building a 1.4.1 release; he has to finish up the zaptel 1.4.1 first; but us flunkies at Digium are holding him up at the moment... ;) So, the answer is soon real soon! murf smime.p7s Description: S/MIME cryptographic signature ___ --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] Asterisk 1.4.1
Forrest Beck wrote: Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... I wouldn't be building anything at all on a production machine without doing some testing on another machine first. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] build rpm fails
Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? Any feedback is appreciated. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote: Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. Well, you'll probably find out at the end that you need to upgrade speex to the version above. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? The above rpms are effectively on CentOS: They were built on RHEL, but CentOS is a clone from RHEL. -- Axel.Thimm at ATrpms.net pgpn5JFoRK0i9.pgp Description: PGP signature ___ --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] Re: build rpm fails
Thanks for saving me the time. I will try and yum from ATrpms. On 3/2/07, Axel Thimm [EMAIL PROTECTED] wrote: On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote: Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. Well, you'll probably find out at the end that you need to upgrade speex to the version above. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? The above rpms are effectively on CentOS: They were built on RHEL, but CentOS is a clone from RHEL. -- Axel.Thimm at ATrpms.net ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium S101i - pickupexten doesn't work
How to configure Digium S101i adapter to work with pickupexten *8 ? I have few Sipura adapters and *8 work OK but my new Digium S101i refuses to cooperate. -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gtalktovoip and Asteirsk
Has anyone managed to get gtalktovoip working at all? If so please explain. http://www.gtalk2voip.com/faq.shtml 2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ? A: This is a major feature of our gateway and it is very easy. oGTalk: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oMSN: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oYahoo: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom reject button
I have users in my dialplan that go from SIP to Cell When they are at their desk and they hit reject call, it goes to the next thing in the dialplan, thus transferring to their cell. Not what they want. Is it possible to change the reject button to make it go to voice mail or a new ext? Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 Call locations
I have a SIP user and a remote IAX device I want both to ring 3 times then if neiter pick up it to go to the next thing in the dialplan. Can you do this from the dialplan or do I need to set a hunt group up? Thanks Jason ___ --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] 2 Call locations
From: Jason Walker [EMAIL PROTECTED] Date: Thu, 01 Mar 2007 18:06:46 -0600 I have a SIP user and a remote IAX device I want both to ring 3 times then if neiter pick up it to go to the next thing in the dialplan. Can you do this from the dialplan or do I need to set a hunt group up? Thanks Jason Dial(SIP/SIPuserIAX/IAXuser,6) Asterisk does not seem to count rings so experiment with timeout. (What other hunt groups are out there to set up?) Yuan Liu ___ --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] TDM400p Loaded only once
Thank you for the answer
after
modprobe wctdm
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm
/proc/zaptel/ (empty)
/usr/src/asterisk/zaptel-1.2.14/xpp/utils/genzaptelconf -l (no result)
On 3/1/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote:
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
This is what you attempt to configure
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
And that is the result.
What do you see on /proc/zaptel ?
What is the output of 'xpp/utils/genzaptelconf -l' from the zaptel
source directory?
This is part of my dmesg
audit(1172747900.510:5): avc: denied { net_bind_service }
for pid=1657
comm=hidd capability=10 scontext=system_u:system_r:bluetooth_t:s0
tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
eth0: no IPv6 routers present
[drm] Initialized drm 1.1.0 20060810
ACPI: PCI Interrupt :01:00.0[A] - GSI 16 (level, low) - IRQ 19
[drm] Initialized r128 2.5.0 20030725 on minor 0
agpgart: Found an AGP 3.0 compliant device at :00:00.0.
agpgart: Device is in legacy mode, falling back to 2.x
agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
audit(1172747921.184:6): avc: denied { getattr } for pid=2323
comm=pam_console_app name=card0 dev=tmpfs ino=7969
scontext=system_u:system_r:pam_console_t:s0-s0:
c0.c255tcontext=system_u:object_r:device_t:s0
tclass=chr_file
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.14
Zaptel Echo Canceller: KB1
Load of zaptel. No load of wctdm in sight.
and finally this is my configuration
fxsks=1-4
loadzone=us
defaultzone=us
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk
On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote: Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't support the name feature, so they've changed my link type). Version: Asterisk 1.2.14 svn rev 48468 Interesting, just finished a long day with a client that switched from 20+ POTS to a T1 using NI2 and CallerID Name (on inbound). So I guess NI2 does support CallerID Name. Asterisk Log: Executing Set(SIP/304-091aafb8, CALLERID(all)=Andrewnn) in new stack (I've replaced the digits with n). PRI debug shows: Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 21 80 nn nn nn nn nn nn nn nn nn nn] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'nn' ] From zapata.conf: callerid=asreceived ;Sangoma A101 port 1 [slot:12 bus:0 span: 1] switchtype=ni1 context=from-zaptel overlapdial=yes facilityenable=yes group=0 signalling=pri_cpe channel = 1-23 Thanks! -- Andrew ___ --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] Help Needed: Can't make local calls on a brand new PRI
You sure that you have local service on the PRI? Maybe it's just an LD PRI? On 3/1/07, Matt [EMAIL PROTECTED] wrote: Yes asterisk was stopped and restarted. ztcfg was not rerun. I've never had to rerun that when I made changes in that file before, but we can try it. MY WISH: TelCo Switchmen could talk intelligently about the protocols used on PRIs! On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: It is currently set to unknown. switchtype=national signalling=pri_cpe pridialplan=unknown Was it originally or did you just change it? Did you stop Asterisk and do ztcfg after making the changes? Thanks, Steve ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Polycom reject button
I have users in my dialplan that go from SIP to Cell When they are at their desk and they hit reject call, it goes to the next thing in the dialplan, thus transferring to their cell. Not what they want. Is it possible to change the reject button to make it go to voice mail or a new ext? I don't think so, only options for the polycoms is to move the buttons around on the phone, not change what they actually do. In my dialplan I have the next priority after the dial cmd going to voicemail and the reject button works as expected there. Maybe use the dial-status variables to send a reject to voicemail and a no-answer to send call to cell phone. You would have to test, but I'm sure the dial-status is different between a reject and a no-answer. Good luck. JR ___ --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] No Caller ID Name PRI NI2
What do you mean by outbound CallerID Name? So that when calling a POTS with CallerID service from telco the Name should show up as you send it? If the answer to the above is yes, then stop trying to do that. It won't work, as the name that the POTS subscriber sees is NOT the one you send, but what the provider of that POTS line sees when the do the lookup on the name that is listed (usualy) with the number received as callerid. On 2/28/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces but i'm unable to figure where is the problem. What I see In case that work: incoming call: [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 54 49 4e 20 46 41 58] Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ] PROTOCOL 1F What I see in case that doesn't work: outgoing call: [28 05 b1 69 6e 66 6f] Display (len= 5) Charset: 31 [ info ] completes traces: working: [ 02 01 da d6 08 02 02 34 05 04 03 80 90 a2 18 03 a9 83 81 1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 54 49 4e 20 46 41 58 1e 02 82 83 6c 0c 21 83 38 31 39 37 38 30 31 32 37 33 70 0b a1 38 31 39 33 34 30 30 39 37 37 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 109 0: 0 N(R): 107 P: 0 76 bytes of data -- ACKing all packets from 106 to (but not including) 107 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=76 Call Ref: len= 2 (reference 564/0x234) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 54 49 4e 20 46 41 58] Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ] PROTOCOL 1F 8B 0001 00 (CONTEXT SPECIFIC [11]) A1 0016 (CONTEXT SPECIFIC [1]) 02 0001 01 (INTEGER: 1) 02 0001 00 (INTEGER: 0) 80 000E 49 4E 46 4F 46 4F 52 54 49 4E 20 46 41 58 (CONTEXT SPECIFIC [0]) [1e 02 82 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 21 83 38 31 39 37 38 30 31 32 37 33] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '8197801273' ] [70 0b a1 38 31 39 33 34 30 30 39 37 37] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8193400977' ] -- Making new call for cr 564 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Q.932 Interpretation component is not handled Handle Q.932 ROSE Invoke component [ Handling operation 0 ] Handle Name display operation Received caller name 'INFOFORTIN FAX' -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) q931.c:3294 q931_receive: call 564 on channel 1 enters state 6 (Call Present) Sending Receiver Ready (110) [ 02 01 01 dc ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 110 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter q931.c:2570 q931_call_proceeding: call 564 on channel 1 enters state 9 (Incoming Call Proceeding) [ 00 01 d6 dc 08 02 82 34 02 18 03 a9 83 81 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 107 0: 0 N(R): 110 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 564/0x234) (Terminator) Message type: CALL
[asterisk-users] How can I use the GET VARIABLE variablename in AGI
Hi,All,
I wang to use AGI in asterisk1.4.
AGI file / myperl.agi
#!/usr/bin/perl
use strict;
..
print STDERR 7. Testing GET VARIABLE...;
print GET VARIABLE EXTEN \\\n;
my $result = STDIN;
checkresult($result);
..
when the agi execute; asterisk conosle show that :
AGI Rx GET VARIABLE EXTEN
AGI Tx 520-Invalid command syntax. Proper usage follows:
AGI Tx Usage: GET VARIABLE variablename
Returns 0 if variablename is not set. Returns 1 if variablename
is set and returns the variable in parentheses.
example return code: 200 result=1 (testvariable)
AGI Tx 520 End of proper usage.
--
I couldn't get the global variable ${EXTEN}, who can told me where is the wrong?
Thanks a lot,
Amy
李君
[EMAIL PROTECTED]
2007-03-02
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Re: [asterisk-users] How can I use the GET VARIABLE variablename in AGI
李君 wrote: Hi,All, I wang to use AGI in asterisk1.4. AGI file / myperl.agi #!/usr/bin/perl use strict; .. print STDERR 7. Testing GET VARIABLE...; print GET VARIABLE EXTEN \\\n; my $result = STDIN; checkresult($result); .. I don't know perl, but try either taking the escaped quotes out or surrounding EXTEN with them. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
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Re: [asterisk-users] Re: build rpm fails
Hi Axel, Everything installed and working well. Thanks very much. Quick question, do you have MySQL support compiled into the rpms? On 3/2/07, Axel Thimm [EMAIL PROTECTED] wrote: On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote: Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. Well, you'll probably find out at the end that you need to upgrade speex to the version above. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? The above rpms are effectively on CentOS: They were built on RHEL, but CentOS is a clone from RHEL. -- Axel.Thimm at ATrpms.net ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Newbie extensions.conf question
I'm still stuck on just exactly where in my extensions.conf file I
should put the code below. I'm running 1.2.14 of asterisk.
Chris Griffin
[EMAIL PROTECTED]
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf
exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)
I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?
Thanks,
Chris Griffin
[EMAIL PROTECTED]
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RE: [asterisk-users] How can I use the GET VARIABLE variablename inAGI
From: Àî¾ý [EMAIL PROTECTED]
Date: Fri, 2 Mar 2007 10:53:04 +0800
Hi,All,
I wang to use AGI in asterisk1.4.
AGI file / myperl.agi
#!/usr/bin/perl
use strict;
..
print STDERR 7. Testing GET VARIABLE...;
print GET VARIABLE EXTEN \\\n;
Why do you want put after variable name? AGI is complaining about
syntax. Try
print GET VARIABLE EXTEN\n;
Yuan Liu
my $result = STDIN;
checkresult($result);
..
when the agi execute; asterisk conosle show that :
AGI Rx GET VARIABLE EXTEN
AGI Tx 520-Invalid command syntax. Proper usage follows:
AGI Tx Usage: GET VARIABLE variablename
Returns 0 if variablename is not set. Returns 1 if
variablename
is set and returns the variable in parentheses.
example return code: 200 result=1 (testvariable)
AGI Tx 520 End of proper usage.
--
I couldn't get the global variable ${EXTEN}, who can told me where is the
wrong?
Thanks a lot,
Amy
¡¡
Àî¾ý
[EMAIL PROTECTED]
2007-03-02
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Re: [asterisk-users] Newbie extensions.conf question
I'm still stuck on just exactly where in my extensions.conf file I should put the code below. Chris Griffin [EMAIL PROTECTED] On Feb 28, 2007, at 9:55 PM, Patrick wrote: On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote: Thanks for the link.. As for Google, I know how to use it. I searched for Sven Slezak's Notify and other variations and got Squat.. Yes I had that too initially. The trick is to remove the 's from Slezak. Then the first link that pops up is the link I gave below. On 2/28/07, Patrick [EMAIL PROTECTED] wrote: On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote: What does this module do? On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote: I've installed Sven Slezak's Notify module. http://mezzo.net/asterisk/app_notify.html Google is your friend. Regards, Patrick ___ --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 ___ --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] UK SIP Gateway
I've been using voiptalk.org for about two years now and have never had any problems. I've been using them for my outgoing business calls for a year and am starting to use them for some incoming calls, which is some indication of my comfortableness with their service. My reluctance to move everything over to VOIP is not the technology - it works - rather it is the issue of who owns my business number. None of the VOIP providers that I've come across in the UK have a clear definitive statement about who owns the number. If I'm going to depend on a particular number I want to know that it is going to be mine no matter what happens to the VOIP supplier (e.g. insolvency, take over/merger, exit out of the market, etc.). ___ --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] UK SIP Gateway
On Fri, 02 Mar 2007 07:26:03 +, George Gardiner [EMAIL PROTECTED] wrote: I've been using voiptalk.org for about two years now and have never had any problems. I've been using them for my outgoing business calls for a year and am starting to use them for some incoming calls, which is some indication of my comfortableness with their service. My reluctance to move everything over to VOIP is not the technology - it works - rather it is the issue of who owns my business number. None of the VOIP providers that I've come across in the UK have a clear definitive statement about who owns the number. If I'm going to depend on a particular number I want to know that it is going to be mine no matter what happens to the VOIP supplier (e.g. insolvency, take over/merger, exit out of the market, etc.). ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone: [EMAIL PROTECTED] Hmmm, that is a very interesting point. Some VoIP providers do say you can transfer them though. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Tesco Internet Phone
On Thursday 01 March 2007 20:33, bails wrote: plug it in a linux box and tell us what it is please, generic-usb-audio or what? Bails Julian Lyndon-Smith wrote: Yeah, that's where firefly comes from, doesn't it. I've got the base station plugged in, and the handset connected to it, but it always says pc unavailable. My system (xp) sees a usb phone for speakers and microphone, but I can't get it to work. Did this go any further. I would be interested in this. Tesco also do a unit which plugs into the ethernet into which you can plug a standard analogue phone. Looking at the Freshtel web site implies it is an IAX device, but wonder what controls which peer it connects to. Does anybody know if I could use it to connect to my own Asterisk server? -- Alan Chandler http://www.chandlerfamily.org.uk ___ --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
