[asterisk-users] Re: Tired of fax calls... :-/
Maxim Vexler wrote: > On 7/6/06, Evert Meulie <[EMAIL PROTECTED]> wrote: >> Hi all! >> >> How do I make Asterisk recognize fax calls and disconnect them? >> >> 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 >> > > NVFaxDetect does just that ;) > > Any why, you might find it more useful to actually receive the fax : > http://www.voip-info.org/wiki/view/NVFaxDetect Is there a way to implement this without recompiling the whole server? 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] Tired of fax calls... :-/
Hi all! How do I make Asterisk recognize fax calls and disconnect them? 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] Re: dnid support?
*bump* Anyone? I still can't find little/no info on DNID... :-/ Regards, Evert Evert Meulie wrote: Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 1 -> ext. 1 913 - 2 -> ext. 2 913-1 & 913-2 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the dialparties.agi script. This script sees and identifies the correct dnid, but I am having some trouble to get the dialplan to act on this value. The info in the Wiki ( http://www.voip-info.org/tiki-index.php?page=DNID ) is not of much help either. Anyone here with any suggestions? 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 ___ --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] dnid
Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 1 -> ext. 1 913 - 2 -> ext. 2 913-1 & 913-2 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the dialparties.agi script. This script sees and identifies the correct dnid, but I am having some trouble to get the dialplan to act on this value. The info in the Wiki ( http://www.voip-info.org/tiki-index.php?page=DNID ) is not of much help either. Anyone here with any suggestions? 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] dnid support?
Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 1 -> ext. 1 913 - 2 -> ext. 2 913-1 & 913-2 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the dialparties.agi script. This script sees and identifies the correct dnid, but I am having some trouble to get the dialplan to act on this value. The info in the Wiki ( http://www.voip-info.org/tiki-index.php?page=DNID ) is not of much help either. Anyone here with any suggestions? 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] Re: dual IP connections
Have you checked http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions Regards, Evert [EMAIL PROTECTED] wrote: Hi all, I would like to know if there is a solution to this question. Scenario: Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these link, i.e. doing a load balancing ? Or just better (in my case) to use only one link, and to use the second link as a backup link in the event the first link went down ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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: Asterisk <-> Skype anywhere/anyhow?
Just wondering... Has someone ever contacted Skype/Ebay and asked them about their point of view/opinion on interfacing with SIP / Asterisk? 8-) Regards, Evert [EMAIL PROTECTED] wrote: I sincerely believe that it's completely non-sense to make a channel for Skype. Skype is a *proprietary* protocol. If they(ebay) don't like the idea of someone messing around their network, they will change the protocol specification, launching a new version, for example, and *all* the work and time spent on this will just going to sink. Probably it is better to loose time with something else. Isamar On Mon, 19 Dec 2005, Luigi Rizzo wrote: On Mon, Dec 19, 2005 at 03:18:18PM -0800, Kerry Garrison wrote: I don't know exactly how it works, but since it appears to just be SIP, I would have to assume a STUN setup. I haven't bothered to sit there and watch the packets go by to see what its doing under the hood. thanks - luigi ___ --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: Asterisk <-> Skype anywhere/anyhow?
As soon as they port it to Gentoo I'll try it out... ;-) Evert Kerry Garrison wrote: Everyone should simply uninstall Skype and switch to the Gizmo project because it interfaces quite nicely with Asterisk. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Monday, December 19, 2005 12:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow? Mark Hulber wrote: The paper is definitely interesting and I commend them for their effort but it doesn't represent a complete understanding of the Skype protocol to the extent that an Asterisk server could speak the Skype protocol. They say that much of the Skype protocol is encrypted and needs to be inferred to this point from the types and locations of messages that are being sent. So despite Skype's popularity they basically have their whole product locked down. It is greatly complex, and it also has a number of "stealth" elements that do nasty things with accepted norms of network etiquette. The bottom line is: Skype *is* evil, and the Asterisk folks, for the most part, have on the white hats of Open Source. IMO we should steer 1000 miles clear of it. Yah, yah, "everyone uses Skype." Well everyone uses Micro$oft, too. That doesn't mean Asterisk should get into bed with them. B. ___ --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] Asterisk <-> Skype anywhere/anyhow?
Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? 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] Re: Does hardware like this exist...?
I found the price. $450 :-/ Kevin P. Fleming wrote: Evert Meulie wrote: That unit looks VERY promising! Thanks! :-) Would anyone happen to know an approx. price for a unit like this? Anyone? I bet the manufacturer of the unit would know a price for it, and it's probably even exact, not approximate :-) Since the manufacturer hasn't posted a price, it's likely that nobody knows 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 ___ --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: Does hardware like this exist...?
That unit looks VERY promising! Thanks! :-) Would anyone happen to know an approx. price for a unit like this? Regards, Evert BJ Weschke wrote: On 12/16/05, Evert Meulie <[EMAIL PROTECTED]> wrote: Hi all! I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if you know what I mean... ;-) You're looking for a USB FXS port. Yes, they do exist. You can take a look at the Astribank-8 from Xorcom (www.xorcom.com). I really don't know how well they work as I haven't any personal experience with their equipment, but they were exhibiting this solution at the last Astricon a few months back. -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ --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] Does hardware like this exist...?
Hi all! I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if you know what I mean... ;-) 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] Re: No application 'MeetMe' for extension
Found it! For some reason [EMAIL PROTECTED] had chosen to build itself without app_meetme.so! After building this module by hand, all worked! :-) Evert Evert Meulie wrote: Read before you reply... ;-) To be 100% clear on zaptel/ztdummy, here's the output of my lsmod: [EMAIL PROTECTED] ~]# lsmod Module Size Used by md5 8001 1 ipv6 240097 16 autofs422085 0 i2c_dev14273 0 i2c_core 25921 1 i2c_dev sunrpc139173 1 ztdummy 7748 0 wctdm 40640 0 wcfxo 16928 0 wcte11xp 30496 0 wct1xxp20768 0 wct4xxp57792 0 tor2 93472 0 zaptel196612 7 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 6081 1 zaptel microcode 11873 0 dm_mirror 28449 0 dm_mod 58949 1 dm_mirror button 10449 0 battery12869 0 ac 8773 0 uhci_hcd 32729 0 ehci_hcd 31813 0 hw_random 9557 0 snd_azx20801 0 snd_hda_codec 75844 1 snd_azx snd_pcm_oss52345 0 snd_mixer_oss 21825 1 snd_pcm_oss snd_pcm91973 3 snd_azx,snd_hda_codec,snd_pcm_oss snd_timer 27973 1 snd_pcm snd56997 6 snd_azx,snd_hda_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 12961 1 snd snd_page_alloc 13641 2 snd_azx,snd_pcm 8139too27329 0 mii 8641 1 8139too ext3 118729 2 jbd59481 1 ext3 ata_piix 13253 3 libata 47901 1 ata_piix sd_mod 20545 4 scsi_mod 116429 2 libata,sd_mod Kunal Parikh wrote: Hi Evert, Do you have the zaptel/ztdummy modules installed ? Kunal On 12/8/05, *Evert Meulie* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi all! I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences... In the log-file I see: Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension (from-internal, 8125, 6) This is when I dial 8125 from extension 125. 8125 is defined in the meetme(-additional).conf And before you ask: yes, ztdummy is loaded... Who has any suggestions? I'm stumped... :-/ Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <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
[Asterisk-Users] Re: No application 'MeetMe' for extension
Read before you reply... ;-) To be 100% clear on zaptel/ztdummy, here's the output of my lsmod: [EMAIL PROTECTED] ~]# lsmod Module Size Used by md5 8001 1 ipv6 240097 16 autofs422085 0 i2c_dev14273 0 i2c_core 25921 1 i2c_dev sunrpc139173 1 ztdummy 7748 0 wctdm 40640 0 wcfxo 16928 0 wcte11xp 30496 0 wct1xxp20768 0 wct4xxp57792 0 tor2 93472 0 zaptel196612 7 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 6081 1 zaptel microcode 11873 0 dm_mirror 28449 0 dm_mod 58949 1 dm_mirror button 10449 0 battery12869 0 ac 8773 0 uhci_hcd 32729 0 ehci_hcd 31813 0 hw_random 9557 0 snd_azx20801 0 snd_hda_codec 75844 1 snd_azx snd_pcm_oss52345 0 snd_mixer_oss 21825 1 snd_pcm_oss snd_pcm91973 3 snd_azx,snd_hda_codec,snd_pcm_oss snd_timer 27973 1 snd_pcm snd56997 6 snd_azx,snd_hda_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 12961 1 snd snd_page_alloc 13641 2 snd_azx,snd_pcm 8139too27329 0 mii 8641 1 8139too ext3 118729 2 jbd59481 1 ext3 ata_piix 13253 3 libata 47901 1 ata_piix sd_mod 20545 4 scsi_mod 116429 2 libata,sd_mod Kunal Parikh wrote: Hi Evert, Do you have the zaptel/ztdummy modules installed ? Kunal On 12/8/05, *Evert Meulie* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi all! I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences... In the log-file I see: Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension (from-internal, 8125, 6) This is when I dial 8125 from extension 125. 8125 is defined in the meetme(-additional).conf And before you ask: yes, ztdummy is loaded... Who has any suggestions? I'm stumped... :-/ Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <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] No application 'MeetMe' for extension
Hi all! I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences... In the log-file I see: Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension (from-internal, 8125, 6) This is when I dial 8125 from extension 125. 8125 is defined in the meetme(-additional).conf And before you ask: yes, ztdummy is loaded... Who has any suggestions? I'm stumped... :-/ 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] Re: Suddenly a problem with outgoing calls made from Cisco phones... - SOLVED
Turns out my VoIP provider made a booh-booh... ;-)
Evert Meulie wrote:
> Hi all!
>
> Quite a mystery. The following happened when I was on holiday, and no one
> else has changed any configs of either Asterisk or the Cisco's in the
> building...
>
> The situation: Incoming works fine on all phones. Outgoing only works from
> non-Cisco phones. When calling from a Cisco phone to an external phone, all
> the Cisco-user hears is a ticking crackle and
> after about a minute the phone disconnects.
>
> A 'sip show channels' reveals the following:
>
> Peer User/ANRCall ID Seq (Tx/Rx) Format
> [VoIP-provider] [ext. number dialed]5b1fe97c04d 00103/0 g729
> [IP of Cisco phone][ID of Cisco] 0002b9a7-4b 00102/00102 ulaw
> 2 active SIP channel(s)
>
> Here g729 pops up, even though I have configured [VoIP-provider] to only
> allow/use ulaw/alaw.
>
>
> asterisk -vvv shows:
>
>
> -- Executing Dial("SIP/[ID of Cisco]-4663", "SIP/[VoIP-provider]/[ext.
> number dialed]") in new stack
> -- Called [VoIP-provider]/[ext. number dialed]
> -- SIP/[VoIP-provider]-77a8 is ringing
> -- SIP/[VoIP-provider]-77a8 answered SIP/[ID of Cisco]-4663
> -- Attempting native bridge of SIP/[ID of Cisco]-4663 and
> SIP/[VoIP-provider]-77a8
> 2005-07-13 10:20:48 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible
> codecs!
> 2005-07-13 10:20:48 NOTICE[23280]: channel.c:1736 ast_set_read_format: Unable
> to find a path from g729 to ulaw
> 2005-07-13 10:20:48 NOTICE[23280]: channel.c:1703 ast_set_write_format:
> Unable to find a path from alaw to g729
> 2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to
> transmit frame type 8, while native formats is 256 (read/write = 4/8)
> 2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to
> transmit frame type 8, while native formats is 256 (read/write = 4/8)
> 2005-07-13 10:20:49 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to
> transmit frame type 8, while native formats is 256 (read/write = 4/8)
> 2005-07-13 10:20:49 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible
> codecs!
>
>
>
> I have in my sip.conf in the [general] section the following:
> disallow=all
> allow=ulaw
> allow=alaw
>
> and no allow/disallows at the phones themselves.
>
>
> This used to work just fine... What could have happened...?
>
>
>
> Regards,
> Evert
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Suddenly a problem with outgoing calls made from Cisco phones...
Hi all!
Quite a mystery. The following happened when I was on holiday, and no one else
has changed any configs of either Asterisk or the Cisco's in the building...
The situation: Incoming works fine on all phones. Outgoing only works from
non-Cisco phones. When calling from a Cisco phone to an external phone, all the
Cisco-user hears is a ticking crackle and
after about a minute the phone disconnects.
A 'sip show channels' reveals the following:
Peer User/ANRCall ID Seq (Tx/Rx) Format
[VoIP-provider] [ext. number dialed]5b1fe97c04d 00103/0 g729
[IP of Cisco phone][ID of Cisco] 0002b9a7-4b 00102/00102 ulaw
2 active SIP channel(s)
Here g729 pops up, even though I have configured [VoIP-provider] to only
allow/use ulaw/alaw.
asterisk -vvv shows:
-- Executing Dial("SIP/[ID of Cisco]-4663", "SIP/[VoIP-provider]/[ext.
number dialed]") in new stack
-- Called [VoIP-provider]/[ext. number dialed]
-- SIP/[VoIP-provider]-77a8 is ringing
-- SIP/[VoIP-provider]-77a8 answered SIP/[ID of Cisco]-4663
-- Attempting native bridge of SIP/[ID of Cisco]-4663 and
SIP/[VoIP-provider]-77a8
2005-07-13 10:20:48 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible
codecs!
2005-07-13 10:20:48 NOTICE[23280]: channel.c:1736 ast_set_read_format: Unable
to find a path from g729 to ulaw
2005-07-13 10:20:48 NOTICE[23280]: channel.c:1703 ast_set_write_format: Unable
to find a path from alaw to g729
2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to
transmit frame type 8, while native formats is 256 (read/write = 4/8)
2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to
transmit frame type 8, while native formats is 256 (read/write = 4/8)
2005-07-13 10:20:49 WARNING[23280]: chan_sip.c:1836 sip_write: Asked to
transmit frame type 8, while native formats is 256 (read/write = 4/8)
2005-07-13 10:20:49 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible
codecs!
I have in my sip.conf in the [general] section the following:
disallow=all
allow=ulaw
allow=alaw
and no allow/disallows at the phones themselves.
This used to work just fine... What could have happened...?
Regards,
Evert
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[Asterisk-Users] Asterisk & Windows Messenger 5: Which is the correct/preferred DTMFmode setting?
Hi all! Who can tell me what the correct/preferred/only DTMFmode setting is for Windows Messenger SIP clients? Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dialing from a website. How to start...?
Thanks for the info! That's exactly the pointed I needed! ;-) (but I'll implement it myself. Cheaper...) ;-) ;-) Greetings, Evert Alistair Cunningham wrote: Evert, The best way to do this is have your PHP code put a control file in the outgoing directory of Asterisk. This then invokes an Asterisk macro that calls the user, then transfers them to the contact. The format of the file is at: http://www.voip-info.org/wiki-Asterisk+auto-dial+out I run a consulting firm doing (amongst other things) Asterisk work. If you're interested, we can install Asterisk, configure it to talk to your telephone system, set up the click to dial, and integrate it with your PHP - email me off list. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Evert Meulie wrote: Hi all! We use a PHP-portal for management of our projects & contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing from a website. How to start...?
Hi all! We use a PHP-portal for management of our projects & contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Showing the name of the country on a Cisco 7960/7912?
Hi everyone! I wonder whether the following would be possible: Can Asterisk show the country from which a call originates on the display, along with the phone number? Regards, Evert Meulie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip_xmit errors...
>ping 0.5.0.4 connect: Invalid argument Nope! ;-) Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of 0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument Who can tell me what causes these, and how to fix it...? Is that a valid IP address? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip_xmit errors...
Hi everyone!
Since yesterday evening I'm getting quite a few of the following
errors(?) from Asterisk:
Oct 26 10:09:38 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:38 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:40 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:40 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:43 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:43 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:49 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:09:49 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:10:58 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:10:58 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:11:00 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:11:00 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:11:03 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:11:03 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:11:10 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:11:10 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:19 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:19 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:21 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:21 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:24 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:24 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:30 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 432) to 0.5.0.4 returned -1: Invalid argument
Oct 26 10:12:30 WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of
0x3ddfa3a8 (len 508) to 0.5.0.4 returned -1: Invalid argument
Who can tell me what causes these, and how to fix it...?
Regards,
Evert Meulie
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[Asterisk-Users] Re: I have Asterisk & Hylafax on a server. What else do I need...?
Is there a page/site where the progress/info on this project is to be found? :-) Regards, Evert Meulie Jon Radon wrote: Right now, you'd need an FXS port and a modem for HylaFax to use. It's not an ideal setup, but more reliable than using an ATA such as the Sipura. Steve Underwood is working on a t38 modem for Asterisk which would interface with HylaFax. This would be the ideal setup. :) On Mon, 25 Oct 2004 10:08:43 +0200, Evert Meulie <[EMAIL PROTECTED]> wrote: Hi everyone! I have an Asterisk server here that also has Hylafax installed on it. What else do I need to have that server send/receive faxes? Regards, Evert Meulie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I have Asterisk & Hylafax on a server. What else do I need...?
Hi everyone! I have an Asterisk server here that also has Hylafax installed on it. What else do I need to have that server send/receive faxes? Regards, Evert Meulie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'asterisk' displayed on my Cisco 7960 & 7912...
Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] let incoming callers contact a certain extension...
Hi everyone! The following: Any calls coming in on extension 12121212 should get a message telling them to dial the extension of the person they are trying to reach, and then press #. The call should then go to the entered extension. This is as far as I got... *** exten => 12121212,1,Wait,1 exten => 12121212,2,Answer exten => 12121212,3,PlayBack(welcome) exten => 12121212,4,PlayBack(if-u-know-ext-dial) exten => 12121212,5,PlayBack(then-press-pound) exten => 12121212,6,DigitTimeout,10 exten => 12121212,7,ResponseTimeout,30 *** Who can help me a little further on the way? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dial '0' for outside line and get a dialtone...
Maurizio Marini wrote: On Friday 17 September 2004 11:43, Evert Meulie wrote: How do I implement this in extensions.conf...? maybe this may help... http://lists.digium.com/pipermail/asterisk-users/2004-February/036737.html Thanks! That works like a charm! The only thing I'd like to do now is NOT having to press 'Dial' on my Cisco 7960 between the '0' and the rest of the number. Any options for that...? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial '0' for outside line and get a dialtone...
Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What does 'Forbidden (From header is not a Trust host or gateway)' mean?
Found it. It's a Micronet-specific error message. So much for standards... :-/ Evert Meulie wrote: From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert";tag=as6e18534e To: Call-ID: [EMAIL PROTECTED] ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: "Evert";tag=as6e18534e To: ;tag=87f2a0d5-13c4-4146e24d-1a7b91cc-796f Call-ID: [EMAIL PROTECTED] ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines -- Got SIP response 403 "Forbidden (From header is not a Trust host or gateway)" back from [SIP server of VoIP provider] Transmitting: ACK sip:[dialled [EMAIL PROTECTED] server of VoIP provider] SIP/2.0 Via: SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b From: "Evert" ;tag=as6e18534e To: ;tag=87f2a0d5-13c4-4146e24d-1a7b91cc-796f Contact: Call-ID: [EMAIL PROTECTED] ext. IP] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Question
Dave Cotton wrote: On Thu, 2004-09-16 at 12:00 +0200, Evert Meulie wrote: Dave Cotton wrote: On Thu, 2004-09-16 at 09:35 +, Murali wrote: Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Check the version of mpg123 is it 0.59r this is the only version that really works. What's wrong with 0.59s? That one seems to work fine as well...8-) If you look at the archives you will find this has been discussed at length. 0.59r works for * 0.59s does not. You want MOH to work you use what works with *. Is it possible to search the archives somewhere online? Downloading all those monthly files in mbox format would be a bit too time-consuming for me... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Question
Dave Cotton wrote: On Thu, 2004-09-16 at 09:35 +, Murali wrote: Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Check the version of mpg123 is it 0.59r this is the only version that really works. What's wrong with 0.59s? That one seems to work fine as well...8-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quality of musiconhold...
Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ftp.digium.com/pub/asterisk/webmin
Hi everyone! Is it safe to use this (old!) webmin module with asterisk 1.0rc2? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec trouble?
Hi everyone! Situation: when I call from cell phone to a asterisk-connected phone, all works fine. When I call from the asterisk-connected phone (a Cisco 7960 SIP) to the cell, the connection gets made, but there is no audio going in either way... Asterisk reports the following: Sep 16 08:27:41 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[360465]: rtp.c:1382 ast_rtp_bridge: codec0 = 268 is not codec1 = 0, cannot native bridge. == Spawn extension (sip, , 1) exited non-zero on 'SIP/105-1559' (123.123.123.123 is the IP of our VoIP-provider, is my cell phone, and 105 is the asterisk-connected phone). Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk does not start...
Thanks, that did the trick! :-) Kinda weird though that the mp3's that actually come with Asterisk don't work correctly 'out of the box'. Or is this a mpg123 bug? Regards, Evert Meulie Andreas Roedl wrote: Hello! Am Dienstag, 14. September 2004 19:21 schrieb Evert Meulie: Found new ID3 Header Remove the ID3 Tag from your Music-On-Hold MP3s with: http://fibiger.org/mp3tag.html Andi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk does not start...
When I do a 'asterisk -vc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-09/14/04-15:40:16, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <[EMAIL PROTECTED]> = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found Asterisk Management interface listening on port 5038 == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 -> 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround]
Re: Re: [Asterisk-Users] how to route these outgoing calls?
Tried that. Now I get: Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: "Evert"@>;tag=as0687982f To: >;tag=87f2a0d5-13c4-4146e66c-1a8baa18-5e5 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via:SIP/2.0/UDP 217.13.2.82:5060;branch=z9hG4bK3dc10bb5 Content-Length:0 You can try this: In your sip.conf add the following entry [yourProvider] type = peer secret = username = host = fromuser = ; some prviders need this parameter fromdomain = ; some prviders need this parameter In your extension.conf add the following entry: exten => _NXXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) This config is only for outgoing calls. On Tue, 14 Sep 2004 10:01:37 +0200, Evert Meulie <[EMAIL PROTECTED]> wrote: Hi everyone! I now have obtained a couple of SIP-accounts at a local VOIP-provider. How do I specify that ALL outgoing calls to _NXXX go out via one of these accounts? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What does 'Forbidden (From header is not a Trust host or gateway)' mean?
From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert";tag=as6e18534e To: Call-ID: [EMAIL PROTECTED] ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: "Evert";tag=as6e18534e To: ;tag=87f2a0d5-13c4-4146e24d-1a7b91cc-796f Call-ID: [EMAIL PROTECTED] ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines -- Got SIP response 403 "Forbidden (From header is not a Trust host or gateway)" back from [SIP server of VoIP provider] Transmitting: ACK sip:[dialled [EMAIL PROTECTED] server of VoIP provider] SIP/2.0 Via: SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b From: "Evert" ;tag=as6e18534e To: ;tag=87f2a0d5-13c4-4146e24d-1a7b91cc-796f Contact: Call-ID: [EMAIL PROTECTED] ext. IP] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong ID going out...
Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXX,1,Dial,SIP/[EMAIL PROTECTED] seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert";tag=as0aca53fa To: Call-ID: [EMAIL PROTECTED] IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868 Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: "Evert";tag=as0aca53fa To: ;tag=87f2a0d5-13c4-4146d5ea-1a4b30eb-3af4 Call-ID: [EMAIL PROTECTED] IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868 Content-Length:0 ( [my IP] is my external IP, [voip IP] is the IP of the SIP server of my VoIP provider. [dialled number] is the number I dialled) I don't see any sign here of the username/password being passed to my provider. is that ok? IMHO I think it should identify me as [username]/[password], instead of 'asterisk' to my VoIP provider. What am I doing wrong...? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to route these outgoing calls?
Hi everyone! I now have obtained a couple of SIP-accounts at a local VOIP-provider. How do I specify that ALL outgoing calls to _NXXX go out via one of these accounts? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird routing(?) problem with 2 Asterisk servers
traceroute A -> B: traceroute to 192.168.2.44 (192.168.2.44), 30 hops max, 38 byte packets 1 192.168.11.1 (192.168.11.1) 1.964 ms 1.181 ms 0.852 ms 2 10.138.3.2 (10.138.3.2) 43.428 ms 49.634 ms 47.601 ms 3 192.168.2.44 (192.168.2.44) 53.440 ms 49.320 ms 48.968 ms traceroute B -> A: traceroute to 192.168.11.6 (192.168.11.6), 30 hops max, 40 byte packets 1 192.168.2.1 (192.168.2.1) 1.873 ms 1.861 ms 2.106 ms 2 10.138.3.3 (10.138.3.3) 45.356 ms 44.139 ms 44.884 ms 3 192.168.11.6 (192.168.11.6) 43.390 ms 43.736 ms 45.823 ms 10.138.3.2-10.138.3.3 is the PPTP connection between both systems. Should bindaddr (iax.conf) or externip (sip.conf) be defined for a setup like this one? Regards, Evert Do you know where it got the 10.138.3.2 IP from? Is it configured anywhere on the server? Do you have externip defined in that config file? Evert Meulie wrote: Hi everyone! situation: Asterisk-server A: 192.168.11.6 Asterisk-server B: 192.168.2.44 server B contains a register => username:[EMAIL PROTECTED] But... when I boot it, I get: Registered to '192.168.11.6', who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The 192.168.2.x & 192.168.11.x networks are fully 'connected' to each other... Who knows the answer...? Regards, Evert Meulie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] weird routing(?) problem with 2 Asterisk servers
Hi everyone! situation: Asterisk-server A: 192.168.11.6 Asterisk-server B: 192.168.2.44 server B contains a register => username:[EMAIL PROTECTED] But... when I boot it, I get: Registered to '192.168.11.6', who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The 192.168.2.x & 192.168.11.x networks are fully 'connected' to each other... Who knows the answer...? Regards, Evert Meulie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk & the Micronet SP5210 anyone?
Hi everyone! Is there a way to let Asterisk connect to/interface with the Micronet SP5210 SIP server ( http://www.micronet.info/Products/voip/SP5210.asp )? It does not support IAX, but maybe there is another way...? Greetings, Evert Meulie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'connecting' voip-numbers to our Asterisk - update
Hi! It turns out my provider uses the Micronet SIP server. Any possibilies to let this one interface with Asterisk? Regards, Evert Evert Meulie wrote: Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten => ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register => :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'connecting' voip-numbers to our Asterisk
Hi! Sample configuration or other documentation from the provider? Hmm, haven't received any! :-/ all I got was username & password... Is there a way (perhaps with sipsak?) to determine what kind of server/system they are running? If their system is not IAX-compatible, what are my options then for routing incoming, outgoing or both via this voip-provider? Greetings, Evert Benjamin on Asterisk Mailing Lists wrote: On Wed, 08 Sep 2004 10:08:00 +0200, Evert Meulie <[EMAIL PROTECTED]> wrote: I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten => ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register => :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? No. First of all, let me ask you this... Are you sure that this provider supports IAX? I am asking because the Cisco 7960 doesn't do IAX, so you wouldn't have been using IAX when connecting directly. Second, if your provider does support IAX, then you will also need to set up a peer for incoming connections and send the calls to your incoming context, like so ... [iaxprovider] type=user username=888 secret=blah host=iax.provider.com qualify=yes disallow=all allow=whatever-codec-they-support context=incoming-from-iaxprovider this may or may not work depending on how your provider will try to connect to you. For example, FWD will always come in as user "iaxfwd", so if you don't define your inbound peer as [iaxfwd] it won't work. Also, some providers use passwords, others use RSA keys. but assuming that the above matches the way in which your provider expects to connect to you, then you will still need an incoming context in extensions.conf named the same way as whatever comes after the "context=" setting. Even that may not be enough depending on how your proider presents the call to you. They may come in using your username or number, but they may as well use an account code or simply "s". You will have to check out the sample configuration or whatever other documentation they provide. The chance is that somebody on this list is using the same provider, so you may tell us what provider you are using and somebody may then share their configuration with you. Also, the Wiki may have a sample configuration for the provider you are using. I always use the IAX debug command on the console to find out how an IAX peer comes in. Simply enter the command "iax2 debug" on the Asterisk console, then make a test call and see what the debug output says. It's pretty self explanatory. Use the command "iax no debug" to turn debugging off again. rgds benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'connecting' voip-numbers to our Asterisk
Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten => ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register => :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk & ISDN-card
Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing works, incoming doesn't...
Addition: the console also has these showing: Jul 29 09:58:06 WARNING[1142106560]: chan_sip.c:612 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Evert Meulie Sent: 28 July 2004 15:28 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Outgoing works, incoming doesn't... Hmm, I get lots of these: to 192.168.2.175:5060 Retransmitting #3 (no NAT): OPTIONS sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.6:5060;branch=z9hG4bK442bde8b From: "asterisk" ;tag=as6496d70e To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 28 Jul 2004 13:44:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 IP 192.168.2.175 is the phone IP 192.168.11.6 is Asterisk (it's not a routing problem, since other phones on the 192.168.2.x IP's do show up as 'OK') Regards, Evert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: 28 July 2004 15:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outgoing works, incoming doesn't... Evert Meulie wrote: > Hi! > > Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip > show peers' gives: > > > Name/usernameHostDyn Nat ACL Mask Port > Status > 105/105 192.168.2.175D 255.255.255.255 5060 > UNREACHABLE > > Is there something wrong with the config on that phone? If so, who can > tell me what? As Asterisk tells you, it's UNREACHABLE from Asterisk. Turn on SIP debug and see what happens - where Asterisk is sending packets and if we get any replies at all. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Access voicemail from Cisco 7960
Thanks for your swift reply! It did help me... kind of... ;) Guess what I had to do to get it working on my system? I had to ADD dtmfmode=inband to my config! 8-) But now I have full access to my mailbox! :) Regards, Evert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen Sent: 28 July 2004 16:50 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Access voicemail from Cisco 7960 On Wed, 28 Jul 2004 14:22:17 +0200, Evert Meulie <[EMAIL PROTECTED]> wrote: > Hi everyone! > > Who can tell me how I can access my voicemail? When I dial the > voicemail on my Cisco 7960 I get access, but when trying to enter my > mailbox number it seems that Asterisk doesn't 'get' any of the keys I > press. DTMF problem perhaps? > > Any suggestions on how/where to fix this...? I had a similar problem. If you look at the console, you'll probably either see it missing digits, or sending too many digits. Even though I was using ulaw as my codec, Asterisk didn't like my specifying dtmfmode=inband. I commented out that line and away it went fine. Here is my current sip.conf for my 7960 which works (connected directly to the Asterisk box) [100] type=friend secret=password username=100 callerid="Leif Madsen" <18924> context=extensions ;dtmfmode=inband qualify=yes nat=no host=dynamic canreinvite=no disallow=all allow=ulaw allow=alaw allow=g729 mailbox=100 HTH, Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing works, incoming doesn't...
Hmm, I get lots of these: to 192.168.2.175:5060 Retransmitting #3 (no NAT): OPTIONS sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.6:5060;branch=z9hG4bK442bde8b From: "asterisk" ;tag=as6496d70e To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 28 Jul 2004 13:44:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 IP 192.168.2.175 is the phone IP 192.168.11.6 is Asterisk (it's not a routing problem, since other phones on the 192.168.2.x IP's do show up as 'OK') Regards, Evert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: 28 July 2004 15:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outgoing works, incoming doesn't... Evert Meulie wrote: > Hi! > > Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip > show peers' gives: > > > Name/usernameHostDyn Nat ACL Mask Port > Status > 105/105 192.168.2.175D 255.255.255.255 5060 > UNREACHABLE > > Is there something wrong with the config on that phone? If so, who can > tell me what? As Asterisk tells you, it's UNREACHABLE from Asterisk. Turn on SIP debug and see what happens - where Asterisk is sending packets and if we get any replies at all. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Access voicemail from Cisco 7960
Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing works, incoming doesn't...
Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/usernameHostDyn Nat ACL Mask Port Status 105/105 192.168.2.175D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the config on that phone? If so, who can tell me what? Regards, Evert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
