Re: [asterisk-users] Need T1 crossover cable?
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? As people answered here you need a crossed cable; Note that T1/E1 cables are different than Ethernet. T1/E1 uses pins 1,2,4,5 (while Ethernet uses 1,2,3,6). A crossed T1 cable is: 1 4 2 5 4 1 5 2 Regards, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phones and corporate phone directory
Hello, A few days ago I've posted two questions about Polycom phones: How to access corporate phone directory from the phone and how to use a conference server with it. After I got zero responses I tried openning a support call in Polycom's site. Here are the replies I got from them: - Corporate directory: They are thinking about it, probably will use LDAP. I was asked to open a feature request to which I got no response yet. - conference server: I asked it because there is a a parameter to set an external conference server, and when I tried it only the initiator was transfered to the server; the other parties were left in hold state. Polycom replied that it is not supported and only the phone's internal conference feature should be used. Regards, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to run ztcfg manually?
On Fri, Oct 26, 2007 at 04:52:07PM -0800, Mojo with Horan Company, LLC wrote: I don't have T1 but it seems that the first time I run ztcfg (or in fact, the zaptel startup script runs it for me) it fails. What distribution is it? RHEL4 / CentOS4 has an early udev version that seems to react quite slowly. For that reason that zaptel init.d script includes a delay loop. In earlier versions it had waited up to 10 seconds for /dev/zap/ctl to appear. In current versions it waits up to 20 seconds, and that number is configurable through /etc/sysconfig/zaptel (or /etc/default/zaptel on Debian). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk canreinvite=yes
Dear all I have small lan and i have configure hardphone with my asterisk with one E1 PSTN line now i have configue to use canreinvite=yes in sip.conf If i user conreinvite=no then my RTP goes throgh asterisk means asterisk come in media path and if i user conreinvite=yes then RTP path would be sip phone to sip phone ??? My all phone in LAN not behind the NAT so guessest me what option would be best for my setup PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response time in Asterisk
In what amount of time does 100 Trying message have to be sent to asterisk? I see asterisk retransmitting the INVITE message multiple times before receiving the 100 Trying message. The INVITEs are retransmitted based on a timer T1, which starts at a default of 500 ms and then exponentially backoffs and caps at 64*T1. The first INVITE retransmission is supposed to happen in 500 ms. However, Asterisk has a minor bug in this place. Asterisk sends the first INVITE retransmission after 1 second instead of 500 ms. This means Asterisk will wait for a second for a response such as 100 Trying before it will start retransmitting the INVITE. Asterisk will retransmit the INVITE after 1, 1, 2, 4, 8, 16, 32 seconds (ideally, this should be 500ms, 1, 2, 4, 8, 16, 32) from the start if it doesn't see a response. Raj --- David Boyd [EMAIL PROTECTED] wrote: On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote: I need to know how fast a sip device needs to respond to an INVITE sip message from asterisk before asterisk retransmits the INVITE message again. Thanks Snip --- 7.2.1 INVITE received When an INVITE request is received by the gateway, a 100 Trying response MAY be sent back to the SIP network indicating that the gateway is handling the call. The necessary hardware resources for the media stream MUST be reserved in the gateway when the INVITE is received, since an IAM message cannot be sent before the resource reservation (especially TCIC selection) takes place. Typically the resources consist of a time slot in an E1/T1 and an RTP/UDP port on the IP side. Resources might also include any quality-of-service provisions (although no such practices are recommended in this document). ** After sending the IAM the timer T7 is started. The default value of T7 is between 20 and 30 seconds. The gateway goes to the 'Trying' state. ** ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk canreinvite=yes
On Sat, 27 Oct 2007, satish patel wrote: My all phone in LAN not behind the NAT so guessest me what option would be best for my setup That depends on what you think the best option for your setup is. :) No, seriously. On the one hand, in a LAN environment, it's probably easier for your phones to pass media peer-to-peer than to bog down your Asterisk box with it. On the other hand, if it's just a handful of phones, it really couldn't possibly make any less of a difference. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 Faxing and Asterisk
Hi, Yes, i have used it for T.38 faxing. Thanks, Vivek On 10/26/07, Nasir Iqbal [EMAIL PROTECTED] wrote: Hi, Have you tried Callweaver http://www.callweaver.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Well, unfortunately for you, that is the exact opposite of the philosophy of the Asterisk codebase. Every attempt is made to genericize the channel driver interface so that you do not need to know the details of the underlying driver. Where we have failed to do so in the past, we are attempting to rectify in current approaches. The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? It would allow for those who want to have this information to get it and act accordingly in the dialplan. I know I have missed this possibility, and instead tried to puzzle together information from DIAL_STATUS and HANGUP_CAUSE. I do agree with the general assumption that the dialplan should be generic, but in reality this is often not the case. You add a SIP-header to tell the client to auto-answer or to change the ring tone, or something like that. I guess that there are similiar ways customize things in IAX or ZAP, and thereby making the dialplans not so generic. Our dialplans often depend heavliy on SIP, but that is of course a result of us working in a SIP-only environment. // T ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
http://www.faqs.org/rfcs/rfc3398.html The conversion is lossy. More than 1 SIP cause code is mapped to a Q.931 cause code (in Asterisk at least). See hangup_sip2cause() in chan_sip.c True. The conversion is lossy in that respect and most of the times semantically incorrect simply because of the fundamental differences between SIP and ISUP. In fact, many new SIP response codes have been defined and will be defined in the future since RFC 3398 was written (http://www.iana.org/assignments/sip-parameters). And as far as I can tell a revision of RFC 3398 is not in works in the IETF. - Raj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Friday, October 26, 2007 11:37 AM To: Asterisk Users Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE Eric ManxPower Wieling wrote: On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? There is an RFC for this. I don't know if Asterisk follows the RFC or not. http://www.faqs.org/rfcs/rfc3398.html The conversion is lossy. More than 1 SIP cause code is mapped to a Q.931 cause code (in Asterisk at least). See hangup_sip2cause() in chan_sip.c Regards, Philipp Kempgen -- 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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nokia E65 SIP/2.0 407 Proxy Authentication Required Problem
Hi friends, We have are getting SIP/2.0 407 Proxy Authentication Required on Invite pakcet once Nokia E65 trying to dial number. But it can recive well from other caller. We tried to disable secrete and it worked fine. But we have lot of users and disabling secrete is risky. Interesting thing is Nokia N95, N80 is working well with the secrete the problem is only with Nokia E65. I will be appreciate if some one can help us to solve this issue. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when using PRI signaling. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Friday, October 26, 2007 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines Michelle Dupuis wrote: I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each of the t1 channels out into individual lines (tied to a specific extension) - so a trunk in and out. Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info across the channels so each side knows what to do? Is there something in the PRI protocol you can point me to for figuring this out? Thanks, MD _ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users That's what the D channel is for. A PRI is a primary rate ISDN. B channels carry voice, D channel handles the information signalling in ISDN. Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need T1 crossover cable?
john beaman wrote: For pinout info, check out: http://www.asteriskdocs.org/cables/ John Beaman Telecom Specialist II Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 10/26/2007 4:01:29 PM Michelle Dupuis wrote: I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, use a T1 crossover(not an ethernet crossover). Lyle - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Whenever you connect two pieces of DTE with a single cable you must have the cable in crossover, so yes. Otherwise they would both attempt to transmit and receive on the same pairs. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chanspy or Extenspy.
Sir, I have configured chanspy and extenspy to listen call on any extension but in both case i am unable to hear voice only silence is there. Rajeev. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
Michelle Dupuis wrote: Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when using PRI signaling. When you are using a PRI, you'll see something like: Accepting call from '248xxx' to '734xxx' on channel 0/1, span 1 So, for the inbound, you'd have an entry that would match against 734. I have the following in my PRI context: exten = _734XXX,1,Gosub(check_blacklist,s,1) exten = _734XXX,2,NoOP(Caller not blacklisted) exten = _734XXX,3,Gosub(get_name,s,1) exten = _734XXX,4,Goto(incoming,s,1) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
The same as any other zap channel does. That is part of the magic of the zaptel drivers. Lyle Michelle Dupuis wrote: Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when using PRI signaling. Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Lyle Giese *Sent:* Friday, October 26, 2007 5:54 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines Michelle Dupuis wrote: I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each of the t1 channels out into individual lines (tied to a specific extension) - so a trunk in and out. Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info across the channels so each side knows what to do? Is there something in the PRI protocol you can point me to for figuring this out? Thanks, MD ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users That's what the D channel is for. A PRI is a primary rate ISDN. B channels carry voice, D channel handles the information signalling in ISDN. Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call center manager for Asterisk (Release 0.5)
CCMANAGER 0.5 released!! NOTE: this is a previous alpha release, maybe there is some customization to do on the settings files, i can't write a clear and complete howto at the moment I don't have released upgrades in the last months but the project is still alive i'm too busy at the moment, i'm following other projects to have some resources (both money and time) and then i can continue this project. Otherwise, i think that new upgrades will follow in the next months, if you have requests post it to the mailinglist on sourceforge I'm still looking to people that want to join this project, the new steps are: - integration with AJAX - project and implementation of an XML layer to manage n server (load balancing, logging and so...) from one ccmanager NEWS: the most important news is that ccmanager reports now supports both the native format that the new reportmaker format ( http://sourceforge.net/projects/reportmaker ) FEATURES: - users management - call generation (making a GET or POST request on a certain URL) - queue management (LOGIN / LOGOUT / QUEUE STATUS) - pickup a call from a queue even if the user isn't logged in the queue - outbound call in customizable context - queue stats import from queue_log - queue reports creation (using an open xml format and reportmaker format) - report export in - html - rtf - xls - pdf FEATURES OF REPORTMAKER reportmaker allows you to define a generic report in xml containing sections,graphs,tables,images. The data can be retrieved directly with sql query. The report can be exported in various formats (html,xml,rtf,pdf) CHANGELOG: 20/08/2007 - added the possibility to specify a different database directly in the report - added the project reportmaker for the report generation - mantained the compatibility with old ccmanager report style - fixed the css for calendar 11/07/2007 - added the file update_stats.php - changed the update method 16/03/2007 - fixed an error for the stats / update script (event ABANDON) - changed the date fromat from Y-m-d h:i:s to Y-m-d H:i:s - addedd the possibility to have multiple graphs on a report - added 6 new reports 14/03/2007 - added the module reports - integrated the module reports with the module stats - now you can generate your reports using an xml format 11/03/2007 - added the module stats - updated the file db.sql with sql instructions for the creation of queue_stats table - added the files view.sql bye -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
Ok - that's great. I see how the destination number will match to the exten value, but how do I access the from number '248xxx'? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, October 27, 2007 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines Michelle Dupuis wrote: Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when using PRI signaling. When you are using a PRI, you'll see something like: Accepting call from '248xxx' to '734xxx' on channel 0/1, span 1 So, for the inbound, you'd have an entry that would match against 734. I have the following in my PRI context: exten = _734XXX,1,Gosub(check_blacklist,s,1) exten = _734XXX,2,NoOP(Caller not blacklisted) exten = _734XXX,3,Gosub(get_name,s,1) exten = _734XXX,4,Goto(incoming,s,1) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unlocking Cisco 7921
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For the purpose of the pilot (i.e. low investment) I want to configure the phones from the keypad. Each phone shows settings locked! whenever I try to edit the network profiles. I can't seem to unlock them! Hopefully there is a secret button combination...I would hate to have to go to a Cisco Unified CallManager just to unlock a few phones... Thanks, MD ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to the exten
value, but how do I access the from number '248xxx'?
exten = s,1,GotoIf($[${CALLERID(number)} = 248xxx ]?2:3)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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Re: [asterisk-users] Unlocking Cisco 7921
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For the purpose of the pilot (i.e. low investment) I want to configure the phones from the keypad. Each phone shows settings locked! whenever I try to edit the network profiles. I can't seem to unlock them! Hopefully there is a secret button combination...I would hate to have to go to a Cisco Unified CallManager just to unlock a few phones... On other Cisco modemls the sequence to unlock them (as long as you did not set a password) is **# (star, star, pound sign). I hope it works on them as well. __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk canreinvite=yes
Alex Balashov wrote: On Sat, 27 Oct 2007, satish patel wrote: My all phone in LAN not behind the NAT so guessest me what option would be best for my setup That depends on what you think the best option for your setup is. :) No, seriously. On the one hand, in a LAN environment, it's probably easier for your phones to pass media peer-to-peer than to bog down your Asterisk box with it. On the other hand, if it's just a handful of phones, it really couldn't possibly make any less of a difference. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Probably best to no allow reinvites if you plan on using features in features.conf and things such as recording, conferencing, call parking, music on hold. A decent server should not get too bogged down by passing RTP streams, especially if they are the same codec (inf act, reinvite will not work unless both devices share the same codec) and you have gigabit. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
On Saturday 27 October 2007 08:14:05 Torbjörn Abrahamsson wrote: Well, unfortunately for you, that is the exact opposite of the philosophy of the Asterisk codebase. Every attempt is made to genericize the channel driver interface so that you do not need to know the details of the underlying driver. Where we have failed to do so in the past, we are attempting to rectify in current approaches. The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory If you read all of what I said, you'd see that it's not contradictory at all. I said that there are places where we've failed to maintain that segregation and that we're working to rectify that, where possible. to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? It would allow for those who want to have this information to get it and act accordingly in the dialplan. I know I have missed this possibility, and instead tried to puzzle together information from DIAL_STATUS and HANGUP_CAUSE. I do agree with the general assumption that the dialplan should be generic, but in reality this is often not the case. You add a SIP-header to tell the client to auto-answer or to change the ring tone, or something like that. I guess that there are similiar ways customize things in IAX or ZAP, and thereby making the dialplans not so generic. Our dialplans often depend heavliy on SIP, but that is of course a result of us working in a SIP-only environment. Not all of us are working in a SIP-only environment. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime on Asterisk 1.2.24
I *STRONGLY* recommend that you do NOT use realtime extensions. If you want a dynamic dialplan, the correct way to do it is to segregate your logic and your data (via something like func_odbc), not to stick all of your logic into a database. -- Tilghman I'm confused by your statement here. Isn't it the whole idea behind ARA to put as much of your dial plan in a database? Or are you referring to static extensions.conf settings? I'm just getting into func_odbc and maybe I'm still not completely clear on the benefits over res_mysql with ARA. 95% of my dial plan is in a database and is working fine for me. Please clarify if you have a moment. Thanks. JR --- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EM.One
Pat Phelan posted this on Facebook - thought the SIP functionality would interest some people here as well. Jajah inks large Japanese dealJajah inks large Japanese dealJajah inks large Japanese deal (Pat Phelan http://www.facebook.com/profile.php?id=754567533 ) http://www.facebook.com/profile.php?id=754567533 By Pat Phelan http://www.facebook.com/profile.php?id=754567533 Interests: pat, pat phelan Share http://www.facebook.com/profile.php?id=674616722 * View original post http://apps.facebook.com/blogfriends/track/441347/674616722 Jajah inks large Japanese deal http://apps.facebook.com/blogfriends/track/441347/674616722 Jajah http://www.jajah.com/ appear to be on a huge push with the recent launch of Jajah buttons and now news coming in from Japan. http://www.flickr.com/photos/[EMAIL PROTECTED]/1755994158/ They have just launched a pre-installed VoIP client for the EM*ONE α device (see attachment) for the Japanese mobile carrier EMOBILE http://www.emobile.jp/ . The software uses the data channel (HSDPA) to deliver voice services for EMOBILE's mobile broadband customers. By running Windows Mobile 6, users can now make native SIP calls using the JAJAH software, to more than 120 global destinations. EMOBILE is a Japanese mobile carrier with a data only connection featuring the EM*ONE α, a mobile device by SHARP. It has a large touch screen and is the perfect symbiosis of a full-fledged Internet browser, a mobile PC and a mobile phone. The EM*ONE α has an HSDPA data connection with 3.6MB high speed internet connectivity. In addition it has a TV antenna and features high resolution TV channels. JAJAH Mobile for the EM*ONE α is a SIP-based VoIP application that ships with the device and connects its users over their mobile broadband Internet connection to regular phones, landline or mobile. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). image001.gifimage002.gifimage003.jpgimage004.jpg___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
Michelle Dupuis wrote: Ok..so how would the CALLED and CALLERID ID be presented to Asterisk when using PRI signaling. The CALLING and CALLED numbers are sent automatically during the call setup. CALLING NAME is usually sent right after the call setup happens. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime on Asterisk 1.2.24
On Saturday 27 October 2007 11:07:32 JR Richardson wrote:
I *STRONGLY* recommend that you do NOT use realtime extensions. If you
want a dynamic dialplan, the correct way to do it is to segregate your
logic and your data (via something like func_odbc), not to stick all of
your logic into a database.
I'm confused by your statement here. Isn't it the whole idea behind ARA to
put as much of your dial plan in a database?
No, the point of ARA is to put stuff into a database, so that it can be
configured via a tool other than a text editor. For some things, notably
channel settings, that goal works very well. For others, such as the logic
of the dialplan, it reduces the efficiency of Asterisk by a large factor.
Or are you referring to
static extensions.conf settings? I'm just getting into func_odbc and maybe
I'm still not completely clear on the benefits over res_mysql with ARA.
95% of my dial plan is in a database and is working fine for me.
The benefit of segregating your logic and your data is to allow extensions to
be created much easier, simply by adding a single row to a database which
contains only the parts of the dialplan which differ. So, for example, a very
simple table might be composed of the following fields:
channel SIP/101 SIP/102SIP/103
extension 101 102
locationcelloffice
cell5551010 5552020
nameRecep Sales
And your extensions.conf might be built along the lines of:
exten = _XXX,1,Set(ARRAY(channel,location,cell)=${ODBC_LOOKUP(${EXTEN})})
exten = _XXX,n,GotoIf($[${LEN(channel)} = 0]?i,1)
exten = _XXX,n,GotoIf($[${location} = cell]?forward_cell)
exten = _XXX,n,GotoIf($[${location} = away]?voicemail)
exten = _XXX,n,Dial(${channel},30)
exten = _XXX,n(voicemail),Voicemail${EXTEN},${IF($[${DIALSTATUS} =
BUSY]?b:u)})
exten = _XXX,n,Hangup
exten = _XXX,n(forward_cell),Dial(Zap/g1/${cell},30)
exten = _XXX,n,Goto(voicemail)
That's just a very brief example of segregating your logic and your data, but
it provides a better way to put the stuff which makes the most sense to be
dynamic in a database and lets everything else (i.e. the logic) sit in a
static configuration file. BTW, I have not included an example
func_odbc.conf, but its contents should be fairly obvious from this example.
--
Tilghman
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[asterisk-users] Little OT: Compilation of EICON driver fails with capi errors
Hello, I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server. The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the kernel finishes without any problems. I have downloaded and installed the deb-source package that EICON/DIALOGIC offers. Th installation script crashes with the following error messages: # LOG START SECTION read kernel version -- #+ LOG INFO: /proc/version: Linux version 2.6.23.1 ([EMAIL PROTECTED]) (gcc version 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2)) #3 SMP Sat Oct 27 11:59:57 CEST 2007 #+ LOG INFO: /etc/*-release: DISTRIB_ID=Ubuntu DISTRIB_RELEASE=7.10 DISTRIB_CODENAME=gutsy DISTRIB_DESCRIPTION=Ubuntu 7.10 #+ LOG INFO: Makefile/VERSION = 2 #+ LOG INFO: Makefile/PATCHLEVEL = 6 #+ LOG INFO: Makefile/SUBLEVEL = 23 #+ LOG INFO: Makefile/EXTRAVERSION = .1 # LOG END SECTION read kernel version -- [ cut ] #+ LOG INFO: end modules_prepare WARNING: Symbol version dump /usr/src/linux-2.6.23.1/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] drivers/isdn/capi/kcapi.o drivers/isdn/capi/kcapi.c: In function ‘recv_handler’: drivers/isdn/capi/kcapi.c:308: warning: format ‘%s’ expects type ‘char *’, but argument 3 has type ‘struct _cdebbuf *’ drivers/isdn/capi/kcapi.c: In function ‘capi_ctr_handle_message’: drivers/isdn/capi/kcapi.c:331: warning: format ‘%s’ expects type ‘char *’, but argument 3 has type ‘struct _cdebbuf *’ drivers/isdn/capi/kcapi.c:354: warning: format ‘%s’ expects type ‘char *’, but argument 3 has type ‘struct _cdebbuf *’ drivers/isdn/capi/kcapi.c: In function ‘capi20_put_message’: drivers/isdn/capi/kcapi.c:671: warning: format ‘%s’ expects type ‘char *’, but argument 3 has type ‘struct _cdebbuf *’ drivers/isdn/capi/kcapi.c:1013:50: error: macro INIT_WORK passed 3 arguments, but takes just 2 drivers/isdn/capi/kcapi.c: In function ‘kcapi_init’: drivers/isdn/capi/kcapi.c:1013: error: ‘INIT_WORK’ undeclared (first use in this function) drivers/isdn/capi/kcapi.c:1013: error: (Each undeclared identifier is reported only once drivers/isdn/capi/kcapi.c:1013: error: for each function it appears in.) drivers/isdn/capi/kcapi.c:1014:47: error: macro INIT_WORK passed 3 arguments, but takes just 2 make[2]: *** [drivers/isdn/capi/kcapi.o] Error 1 make[1]: *** [drivers/isdn/capi] Error 2 make: *** [_module_drivers/isdn] Error 2 #+ LOG INFO: pwd:/usr/lib/eicon/divas/src #! LOG ABORT EXECUTION DUE TO ERROR : Failed to call 'make modules' #! LOG ERROR INFO: make modules Has anyone on the list experienced similar problems with an EICON card and has found a solution? I also tested the installation with kernel 2.6.22-14 which is the one that comes with Ubuntu - same problem. I already contacted EICON support but they haven't answered yet. Thanks for any hint. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime on Asterisk 1.2.24
Tilghman Lesher wrote: I *STRONGLY* recommend that you do NOT use realtime extensions. If you want a dynamic dialplan, the correct way to do it is to segregate your logic and your data (via something like func_odbc), not to stick all of your logic into a database. Should this be taken as a warning to us happy realtime users that it is deprecated, and/or likely to go away? If that's where it's headed (once upon a time a discussion to do it the right way was scheduled for the Atlanta confab last spring, but the topic never emerged there AFAIK) then it ought to be officially deprecated so those of us who use it extensively can begin to plan our migration to other ways of solving the problems that realtime seems (at least to me) to solve nicely. I can deploy large numbers of servers with complex and coherent dialplans with pretty much zero effort on a given new client, and I can also effect system-wide changes across my servers with a single database update. I find it to be a powerful and useful feature. But if there's a better way to do it I'm willing to learn. To my knowledge there are no Postgres ports of ODBC running yet under openWRT, and so I am using the PG module to access my information. At the moment func_odbc wouldn't seem to get the job done for me as per your suggestion above. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.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: encryption support
Russell Bryant wrote: Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.13 and I need to use encryption among Asterisk and my SIP users, and with the RTP data interchanged among users. I prefer the use of ZRTP/SRTP because we use Twinkle and X-Lite/Zfone as our voip clients and they support these encryption mechanism. My question is: do I have to enable any encryption support in Asterisk 1.4.13 ??? Or Asterisk has native encryption support ??? The only VoIP encryption provided in Asterisk 1.4 is IAX2 encryption, which can be used between Asterisk servers. I'm not aware of any IAX2 clients that support encryption. What's the status of SRTP? I remember seeing things floating around about it being under development, but various sotto voce conversations I've had around over the past few days would indicate that it hasn't gained much/any traction. I'd be glad to be disabused of that notion. Thx. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime on Asterisk 1.2.24
Or just generate config files (or parts of config files) from a database dynamically. On Sat, 27 Oct 2007, Brian Capouch wrote: Tilghman Lesher wrote: I *STRONGLY* recommend that you do NOT use realtime extensions. If you want a dynamic dialplan, the correct way to do it is to segregate your logic and your data (via something like func_odbc), not to stick all of your logic into a database. Should this be taken as a warning to us happy realtime users that it is deprecated, and/or likely to go away? If that's where it's headed (once upon a time a discussion to do it the right way was scheduled for the Atlanta confab last spring, but the topic never emerged there AFAIK) then it ought to be officially deprecated so those of us who use it extensively can begin to plan our migration to other ways of solving the problems that realtime seems (at least to me) to solve nicely. I can deploy large numbers of servers with complex and coherent dialplans with pretty much zero effort on a given new client, and I can also effect system-wide changes across my servers with a single database update. I find it to be a powerful and useful feature. But if there's a better way to do it I'm willing to learn. To my knowledge there are no Postgres ports of ODBC running yet under openWRT, and so I am using the PG module to access my information. At the moment func_odbc wouldn't seem to get the job done for me as per your suggestion above. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? I'd agree that SIP response code should be accessible from the dial plan. Knowing the exact SIP response code could be critical for making call processing decisions. The conversion of SIP response codes to Q.931 codes (HANGUPCAUSE) is just too lossy. Building a truly protocol agnostic dial plan API is a worthy goal. But, I think it is somewhat of an unsolvable problem. The signaling protocols are very different and for various reasons people have always wanted access to native information elements carried in the protocol. Perhaps, a very simple solution for this problem could be to support a keyword such as TOPLINE in the SIP_HEADER function to fetch the topmost line in a SIP message. This will not only get the caller the response code for SIP response messages, but will also have the nice byproduct of making the Request-URI available if the message in question is a SIP request. - Raj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Little OT: Compilation of EICON driver fails with capi errors
On Sat, 2007-10-27 at 19:49 +0200, Stefan Guenther wrote: Hello, I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server. The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the kernel finishes without any problems. I have downloaded and installed the deb-source package that EICON/DIALOGIC offers. Th installation script crashes with the following error messages: [snip] I have a couple of single BRI and one quad BRI Eicon Diva Server cards. On Fedora 6 and 7 and CentOS 4.x and 5 there is no need to install anything from Eicon. The kernel already includes the modules for these Eicon Diva Server cards. Here is how I load the modules manually: /sbin/modprobe -v divas /sbin/modprobe -v diva_idi /sbin/modprobe -v kernelcapi /sbin/modprobe -v capi /sbin/modprobe -v divacapi Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime on Asterisk 1.2.24
On Saturday 27 October 2007 13:10:37 Brian Capouch wrote: Tilghman Lesher wrote: I *STRONGLY* recommend that you do NOT use realtime extensions. If you want a dynamic dialplan, the correct way to do it is to segregate your logic and your data (via something like func_odbc), not to stick all of your logic into a database. Should this be taken as a warning to us happy realtime users that it is deprecated, and/or likely to go away? No, it's not deprecated. That said, there are things that are unlikely to ever be supported in realtime extensions, such as hints. What you have now is likely the most that you'll ever have, with regard to realtime extensions, unless somebody contributes some code that surmounts the (fairly considerable) engineering issues. If that's where it's headed (once upon a time a discussion to do it the right way was scheduled for the Atlanta confab last spring, but the topic never emerged there AFAIK) then it ought to be officially deprecated so those of us who use it extensively can begin to plan our migration to other ways of solving the problems that realtime seems (at least to me) to solve nicely. func_odbc (and to a lesser extent, the MYSQL command) seems to be the way forward. I can deploy large numbers of servers with complex and coherent dialplans with pretty much zero effort on a given new client, and I can also effect system-wide changes across my servers with a single database update. There are other ways to do that, such as storing your configuration files in SVN, then doing an '#exec' out of a static file to 'svn cat' the latest version. I find it to be a powerful and useful feature. But if there's a better way to do it I'm willing to learn. To my knowledge there are no Postgres ports of ODBC running yet under openWRT, and so I am using the PG module to access my information. At the moment func_odbc wouldn't seem to get the job done for me as per your suggestion above. Embedded systems are a challenge all of the way around. As I'm not working directly on any embedded systems, I haven't really gone into those issues (other than the autoconf issues you've complained about on this list, and I subsequently fixed). -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Treating T1 as trunk in/out, not individual lines
DL == Doug Lytle [EMAIL PROTECTED] writes:
DL Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to
the exten value, but how do I access the from number '248xxx'?
DL exten = s,1,GotoIf($[${CALLERID(number)} = 248xxx ]?2:3)
That works, of course, but there's also the traditional
ex-gf-function:
exten = s/248XXX,1,NoOp(Matched 248...)
exten = s/X!,1,NoOp(Didn't match)
/Benny
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Re: [asterisk-users] Unlocking Cisco 7921
That did the trick! It appears that all of the config is retrieved from a .cnf.xml file, so there wasn't much more I could do at the phone level other than set the networking parameters. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yehavi Bourvine +972-8-9489444 Sent: Saturday, October 27, 2007 11:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Unlocking Cisco 7921 I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For the purpose of the pilot (i.e. low investment) I want to configure the phones from the keypad. Each phone shows settings locked! whenever I try to edit the network profiles. I can't seem to unlock them! Hopefully there is a secret button combination...I would hate to have to go to a Cisco Unified CallManager just to unlock a few phones... On other Cisco modemls the sequence to unlock them (as long as you did not set a password) is **# (star, star, pound sign). I hope it works on them as well. __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to combine a Fritz ISDN card with analogue handsets
I want to use a Fritz AVM ISDN card to create a switch which is connected to 4 analogue extensions. I believe I need a 4 port FXS module for that, are there any cheap but reliable options out there? Are there some guides that go through the whole process? /voipfc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Display name when dialing on Polycom
I have a customer who wants the Polycoms to display the CallerID name of the person they called on the phone they are calling from. The receiving phone gets CID just fine, but the calling phone doesn't display a name. For instance, if you dialed extension 3000, the Polycom Displays 3000(3000) Instead of John Doe (3000). How do I set that up? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Provisioning Tool Update
I have added directory creation support from CSV as well as a bug fix. V0.0.3 is available http://www.wintrisk.com/ppt.html Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Little OT: Compilation of EICON driver fails with capi errors
On Sat, 27 Oct 2007, Stefan Guenther wrote: Hello, I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server. The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the kernel finishes without any problems. I have downloaded and installed the deb-source package that EICON/DIALOGIC offers. Th installation script crashes with the following error messages: ... drivers/isdn/capi/kcapi.c:1014:47: error: macro INIT_WORK passed 3 arguments, but takes just 2 make[2]: *** [drivers/isdn/capi/kcapi.o] Error 1 make[1]: *** [drivers/isdn/capi] Error 2 make: *** [_module_drivers/isdn] Error 2 #+ LOG INFO: pwd:/usr/lib/eicon/divas/src #! LOG ABORT EXECUTION DUE TO ERROR : Failed to call 'make modules' #! LOG ERROR INFO: make modules Has anyone on the list experienced similar problems with an EICON card and has found a solution? I also tested the installation with kernel 2.6.22-14 which is the one that comes with Ubuntu - same problem. I already contacted EICON support but they haven't answered yet. It looks like the kcapi module that's coming with the Eicon package is incompatible with the new kernel version. If you don't have the need for using the Eicon package, you might want to try the Melware V3 driver which uses the in-kernel capi and will not patch the kernel on your system. Armin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Little OT: Compilation of EICON driver fails with capi errors
On Sat, 27 Oct 2007, Patrick wrote: On Sat, 2007-10-27 at 19:49 +0200, Stefan Guenther wrote: Hello, I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server. The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the kernel finishes without any problems. I have downloaded and installed the deb-source package that EICON/DIALOGIC offers. Th installation script crashes with the following error messages: [snip] I have a couple of single BRI and one quad BRI Eicon Diva Server cards. On Fedora 6 and 7 and CentOS 4.x and 5 there is no need to install anything from Eicon. The kernel already includes the modules for these Eicon Diva Server cards. Here is how I load the modules manually: ... Yes, if you don't any of the new features, you can go with the V2 driver, which is part of the kernel. But newer cards need the new driver. Armin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Uniden UIP200 phones
I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) But it doesn't give me the ring I want on the phone. I have firmware BS4.63 and BS4.77 on the phones and it doesn't seem to work on either. Any suggestions? Thanks, Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Display name when dialing on Polycom
Michael Munger wrote: I have a customer who wants the Polycoms to display the CallerID name of the person they called on the phone they are calling from. You'll want to see this bug: http://bugs.digium.com/view.php?id=8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registration of Snom 320 phone with Asterisk 1.4.13
Hello, I am experiencing difficulty registering my Snom 320 phone with Asterisk 1.4.13, and have been receiving the same transport error messages on the phone as described in this forum post: http://forums.digium.com/viewtopic.php?p=40554highlight=sid=b6d7fd216103dcdafb0b995aff03f07f Are there any known solutions to this? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Read back of caller ID
I've been looking around for an example of a method of reading back a caller ID value, but I haven't found anything that doesn't use Festival. I'd rather not resort to the Mr. Roboto voice if I can avoid it. Playback of the numbers one at a time is perfectly fine, so I'd like to use the default female Asterisk voice (the sound files are in place on my server). Does anyone have an example of how to accomplish this? Thanks in advance! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read back of caller ID
On Saturday 27 October 2007 11:19:05 pm arkda wrote:
I've been looking around for an example of a method of reading back a
caller ID value, but I haven't found anything that doesn't use Festival.
I'd rather not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a time is perfectly fine, so I'd like to use
the default female Asterisk voice (the sound files are in place on my
server). Does anyone have an example of how to accomplish this?
Thanks in advance!
SayDigits(${CALLERID(num)})
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Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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