[asterisk-users] Use one ring-group for ISN truncs

Hi, A question. We are using TE420 cards. Normally we configure for each truncs one ring-group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more ring-groups to one ring-group?

Re: [asterisk-users] Use one group for ISN truncs

Hi, A question. We are using TE420 cards. Normally we configure for each truncs one group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more groups to one group? Example: Group 1

Re: [asterisk-users] Use one group for ISN truncs

On Mon, Jun 28, 2010 at 09:16:37AM +0200, Arjan Kroon | Mobillion wrote: Hi, A question. We are using TE420 cards. Normally we configure for each truncs one group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel =

[asterisk-users] Problem attended transfer with ilbc

I have an Asterisk server on our LAN that serves our office VOIP phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are ulaw/alaw We use attended transfer extensively. If our trunk is ulaw/alaw they work fine. If the trunk is ilbc we have problems 1- incoming PSTN call routed via

Re: [asterisk-users] Problem attended transfer with ilbc

On Mon, Jun 28, 2010 at 5:15 AM, John Taylor j...@vetsurgeon.org.uk wrote: Any idea what may be happening? acknowledged https://issues.asterisk.org/view.php?id=16287 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)

[asterisk-users] sip add header

It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER=Alert-Info: Ring Answer (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I call the SIP phone directly - however - when I first

Re: [asterisk-users] sip add header

On 28 Jun 2010, at 13:08, Jerry Geis wrote: It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do anything. Are you adding the header before or after you dial the local channel? S --

Re: [asterisk-users] Call drops on group paging asterisk - 1.4.22.1

Thanks Mike! We are using one Aastra phone with expansion module and the remaining 27 phones are from Yealink (new phones that came out), currently Aastra phone used to freeze while paging, but now we replaced the aastra to Yealink and will see if this solves the problem. Sandesh On Fri, Jun

Re: [asterisk-users] Problem attended transfer with ilbc

Paul Belanger schrieb: On Mon, Jun 28, 2010 at 5:15 AM, John Taylor j...@vetsurgeon.org.uk wrote: Any idea what may be happening? acknowledged https://issues.asterisk.org/view.php?id=16287 hello, i´ve reported the same bug i´ve found out later with this issue:

[asterisk-users] Pickup a ringing Queue member

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192type=bug' -O - | patch -p0/* ?? Does this mean I have a

Re: [asterisk-users] Pickup a ringing Queue member

On Mon, Jun 28, 2010 at 10:00 AM, Jonas Kellens jonas.kell...@telenet.be wrote: I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing :  wget

Re: [asterisk-users] Pickup a ringing Queue member

Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Yes. What is then the unpatched version of Asterisk 1.4.30 ?? Jonas. -- _ -- Bandwidth

Re: [asterisk-users] Pickup a ringing Queue member

On 28 Jun 2010, at 15:36, Jonas Kellens wrote: Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Yes. What is then the unpatched version of Asterisk 1.4.30 ?? The one you have before you apply the patch?.. --

[asterisk-users] Never seen Problem !!!

One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status

[asterisk-users] Handling DTMF for number 4

Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK

Re: [asterisk-users] Handling DTMF for number 4

Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is

Re: [asterisk-users] Handling DTMF for number 4

Thanks Gareth, when you say that I can choose INBAND for DTMF MODE in the GSM Gateway, that implies that the DTMF MODE of the Asterisk extension registered for the GSM Gateway has to be set to INBAND too or can it remain in RFC2238 ??? Because I have all my Asterisk extensions and IP telephones

Re: [asterisk-users] Handling DTMF for number 4

It would need to be set in the gsm gateway and in the corresponding section in sip.conf which connects to that gsm gateway. Everything else should be left as rfc. It may help or it might not. The gateway might also have some settings you can change to improve the detection. The Patton unit I

Re: [asterisk-users] Never seen Problem !!!

Let me know if you need any further info !! On Mon, Jun 28, 2010 at 9:15 PM, G M gm.cu...@gmail.com wrote: One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by

[asterisk-users] sip server

Hi, Can i use asterisk as sip server for manage call Transmission between gateways Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] sip server

Yes CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mohamed daif Sent: Monday, June 28, 2010 2:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip server Hi, Can i use asterisk as sip server for

[asterisk-users] Asterisk 1.6 and multiple parking

Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same

Re: [asterisk-users] sip server

hi i want to use asterisk as a sip server without installing any hardware in this machine the question is how can i configure the external getaways with asterisk how can i configure the costumer who is i provide calls to hem what is the billing software can i use to calculate the the calls and

[asterisk-users] Problem with TE411P and DAHDI

We just recently upgraded a server from Zaptel to DAHDI and Asterisk 1.4.30 to 1.6.2.9 and now we are getting this message before the server reboots every few minutes: Message from syslogd@ at Mon Jun 28 15:17:48 2010 ... pbx kernel: Dazed and confused, but trying to continue Jun 28

Re: [asterisk-users] sip server

On Mon, Jun 28, 2010 at 4:02 PM, mohamed daif mohamed.d...@gmail.com wrote:  i want to use asterisk as a sip server without installing any hardware in this machine the question is  how can i configure the external getaways with asterisk  how can i configure the costumer who is i provide calls

Re: [asterisk-users] Asterisk 1.6 and multiple parking

Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users]

Re: [asterisk-users] Problem with TE411P and DAHDI

On 06/28/2010 03:27 PM, Carlos Chavez wrote: We just recently upgraded a server from Zaptel to DAHDI and Asterisk 1.4.30 to 1.6.2.9 and now we are getting this message before the server reboots every few minutes: Message from syslogd@ at Mon Jun 28 15:17:48 2010 ... pbx kernel: Dazed

Re: [asterisk-users] sip add header

At 8:08 AM on 28 Jun 2010, Jerry Geis wrote: It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER=Alert-Info: Ring Answer (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I

Re: [asterisk-users] Call file structure and syntax

Well, I¹ve tried this, and something just isn¹t right. Here¹s the context from dialplan show, so I know it¹s loaded anyway: [ Context 'PressTwo' created by 'pbx_config' ] '*' =1. Goto(accept|s|1) [pbx_config] '1' =1.

[asterisk-users] restricting sip users to a certain useragent

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM Customer Relations Management with an SIP dialers built in.the question is the following.. is it possible to force the

Re: [asterisk-users] Call file structure and syntax

Well, I¹ve tried this, and something just isn¹t right. Look here: Event: Hangup Channel: SIP/ShoreTel-1-0004 Cause: 17 Cause-txt: User busy -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Update the LCD with the callee's name after dialing

Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this,

Re: [asterisk-users] restricting sip users to a certain useragent

This is a very good question. I faced the same problem some time ago, and by goggling found out that somebody had actually programmed a patch for this purpose, but it never got approved to go into the main branch of Asterisk. If you google, you'll probably found out details on it. I am, however,

Re: [asterisk-users] Update the LCD with the callee's name after dialing

Remote Party ID in trunk, it works There are hacks for other versions. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux

[asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9

hi, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan --

[asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?

Hi Everyone, I want to know a bit about the guts of the current AsterisNOW system. I know that FreePBX is embraced as the main GUI but is just an install of CentOS 5.4 + (Asterisk/FreePBX from Yum repos)? - Or is there anymore to this? Maybe some security tools? - Or is Asterisk built from the

[asterisk-users] How to configure key sequence in features.conf.

Hi all, I need to achieve the following function: user 1 call to user 2, In the process they calling, if user 2 press *3 keys, then the call hangup and playback voice file. My setting as following: * features.conf** [featuremap] textkey1 = *3