[Asterisk-Users] exten=>h and ResetCDR
Hi friends, I have the entry exten => h,Hangup in my extensions.conf, and I am trying to record the call details for billing. From the wiki i found out that the use of "exten=>h,..." is not suggested for the CDRs. What impact will the use of 'h' make on CDRs? Also, what is the advantage of using ResetCDR with exten=>h? Regards... Girish _ Easiest Money Transfer to India. http://go.msnserver.com/IN/41490.asp Send Money To 6000 Indian Towns. ___ 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] Capabilites of Asterisk
Yes you can connect inbound and outbound to a VOIP carrier and completely forgo the need for any t1/analog cards. Check out www.nuphone.net, and voicepulse. In addition you can trunk together two asterisk boxes on disparate networks and split up the extension mapping, this may be somewhat inefficient though depending on how you handle it. I think the wiki tackles these questions in a bit more depth. Matt On Jan 23, 2004, at 11:35 PM, Ralph Blach wrote: From what I can see, the asterik/digum software/hardward allows for a incomming alalog trunk lines and a lode of internal pbx pots lines. 1)instead of an analog trunk or a t1 trunk, can I connect to a VOIP carrier? 2)can I have a split pbx using VOIP, so that if I have two separate locations it looks like one PBX? Thanks Chip ___ 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 + ADPCM
Checked the archives. I cannot get ADPCM to work with SIP. Calling from phone1 (adpcm) to phone 2(ulaw). Both phones Grandstreams with one set with G726-32 with v0.7.1 cvs. Has anyone got adpcm to work? Jan 24 09:00:14 WARNING[409617]: rtp.c:1069 ast_rtp_write: Not sure about sending format ADPCM packets Jan 24 09:00:14 WARNING[409617]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp format for codec format ADPCM Master ___ 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] Problem installing Asterisk with Mandrake 9.1
Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as if I try and insmod it again, I get: [EMAIL PROTECTED] libpri]# insmod zaptel Using /lib/modules/2.4.21-0.13mdk/misc/zaptel.o insmod: a module named zaptel already exists After a reboot, I attempted to compile LibPri, but I get the error below. Can anyone point me in the right direction? I've tried to repeat the procedure from scratch, checked that I have ncurses, sox, linux source code, openSSL development libraries I need installed. I've re-connected to CVS and gotten the source again, but I get the same results. I'd appreciate any pointers that you may be able to give me as I don't really know where else to look. Thanks, Mike - [EMAIL PROTECTED] libpri]# make install cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o q931.o q931.c ar rcs libpri.a pri.o q921.o prisched.o q931.o ranlib libpri.a cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o pri.lo -c pri.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o q921.lo -c q921.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o prisched.lo -c prisched.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o q931.lo -c q931.c cc -shared -Wl,-soname,libpri.so.1 -o libpri.so.1.0 pri.lo q921.lo prisched.lo q931.lo /sbin/ldconfig -n . ln -sf libpri.so.1 libpri.so mkdir -p /usr/lib mkdir -p /usr/include install -m 644 libpri.h /usr/include install -m 755 libpri.so.1.0 /usr/lib ( cd /usr/lib ; ln -sf libpri.so.1 libpri.so ) install -m 644 libpri.a /usr/lib /sbin/ldconfig [EMAIL PROTECTED] libpri]# ___ 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] Capabilites of Asterisk
From what I can see, the asterik/digum software/hardward allows for a incomming alalog trunk lines and a lode of internal pbx pots lines. 1)instead of an analog trunk or a t1 trunk, can I connect to a VOIP carrier? 2)can I have a split pbx using VOIP, so that if I have two separate locations it looks like one PBX? Thanks Chip ___ 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] MGCP w/8x8 DTA-310 and as5300 pstn gateway
- Original Message - From: "Jeremy Jones" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 12:06 PM Subject: RE: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway >This one does mgcp... It's been used in conjunction with a hosted pbx >system called Centile that 8x8 now owns. If there's a firmware image >anyone knows of to make these do sip, I'd rather do that. But for now, >mgcp help is what I need. You can get the latest SIP firmware from Packet8's TFTP server at 4.42.235.170 file name "current". Read more about it here http://web.packet8.net/download/ Only problems is, in this version the advanced configuration page with the SIP setup is password protected. If you look at the downloaded file, you can see all the HTML stuff for the configuration pages. It may be possible to figure out or remove the password protection. The other option is to load an older version of the SIP firmware in which the SIP page is not protected. I'm sure someone has a copy of it. By the way, do you have a copy of the MGCP firmware in case you want to go back to it? ___ 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] Is it possible to push the media processing off to a gateway for processing?
I was wondering if it is possible to have Asterisk push the media processing off to something with DSP's such as a gateway? That way, asterisk just has to handle the call setups and tear downs. Todd Wallace You mean, like what SIP does by default? This is an incomplete question. Please be more specific. If I have a "gateway", and I have SIP calls coming in from desktop SIP UA's (hardphones or softphones) then Asterisk can simply re-direct those calls to the gateway. Of course, Asterisk _is_ a gateway, so unless you have specific reasons for doing so, it would make more sense to use Asterisk to tackle those jobs with generic, cheap processing horsepower rather than expensive, proprietary DSP's. If you're just getting Asterisk to handle call setups and teardowns, why not just use a real SIP proxy for that? Or do you not know enough about your question to understand why I would differentiate between the two? (not being nasty here, just wondering if I need to explain more) JT ___ 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] RFC3389 support issue with DG104S
On Fri, 2004-01-23 at 17:49, Brian West wrote: > Its silence supression. Turn that off and it will stop doing that. > Before I got my mail in I google search for "RFC3389 dlink" and found the FAQ on http://www.fnords.org/~eric/asterisk/faq.html Q20: I'm getting the message: NOTICE[20498]: File rtp.c, Line 217 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible A20: Asterisk is complaining about silence supression. It's a harmless message, but if you can turn off silence suppression in your SIP client the message will go away. FUnny I missed that page on my original searching :) Once I skipped that "error," I noticed the context error buried in the mounds of messages. Perhaps the "error" should be reworded to a warning, or mention it unimportance. -- Zot O'Connor <[EMAIL PROTECTED]> White Knight Hackers, Inc. ___ 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 + BudgeTone (behind NAT)
At 9:17 PM -0500 1/23/04, Owen Kelso wrote: I've been following up on my problem, which I previously described as: I've concluded that the Netgear router (FVS318) performing the NAT is corrupting the outgoing RTP packets. Traces confirmed that the BudgeTone is sending them out with a UDP checksum of 0 but the next hop after the Netgear router they are set to a non-zero value (an incorrect one). Asterisk is never even seeing the packets because the kernel is recognizing them as corrupt and dropping them, hence the recvfrom() "Resource temporarily unavailable" errors in rtp.c. Here is Netgear's response: Original Message Subject: RE: Webform contact request [#] From:[EMAIL PROTECTED] Date:Fri, January 23, 2004 7:36 pm To: -- SIP VOIP phones do not work with netgear routers. The router will always set a value in the checksum. Regards Netgear Support [EMAIL PROTECTED] Please help us serve you better by clicking here mailto:[EMAIL PROTECTED] if you would like to provide any other valuable feedback. (Note: this feedback is not sent to an agent so you will not receive a reply.) -- Not exactly what I call stellar customer service! I realize this may not be the best solution -- actually, it's probably not even a good solution -- but has anyone experimented with using the Linux SO_NO_CHECK setsockopt() option? It looks like it could be used to ignore the checksums for the RTP packets. Owen Time to dump the Netgear router. That's an unacceptable answer for a router vendor to say "Oh, well, for this MAJOR protocol we're going to simply corrupt those packets so they're unusable." What!? JT ___ 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] RFC3389 support issue with DG104S
At 5:35 PM -0800 1/23/04, Zot O'Connor wrote: I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <[EMAIL PROTECTED]> White Knight Hackers, Inc. Zot - Good to see you're getting around to installing * finally! The error messages you're seeing are probably unavoidable, as many ATA devices do not allow the user to turn on/off the VAD comfort noise stuff. I would suggest that an ugly method to solve the problem is just to find that debug line in the source code and comment it out, since you will have no significant degradation during calls, but you'll see a lot of noisy error messages. JT ___ 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] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
Iain - Brian I believe is correct, and Kannaiyan perhaps is not correct. Perhaps you can post the actual values in one of your call spool files so that we can comment on it more clearly. Using the "Application:" statement in an outbound spool file will prevent a CDR from being created; use "Context:/Extension:/Priority:" methods. If that fails, then we have a bug. JT At 5:59 PM -0600 1/23/04, Brian West wrote: NO it will log from a spool file if and only if you ref an extension and not an application. bkw On Fri, 23 Jan 2004, Kannaiyan Natesan wrote: There is no CDR for the call from spool outgoing, You need to write a patch to solve the same. Kannaiyan - Original Message - From: "Iain Stevenson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 8:27 PM Subject: [Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing? > > I've just noticed that if you start a call by writing a file to > /var/spool/asterisk/outgoing the cdr created on termination logs the call > placed to the local extension - not to the destination in the PSTN. Hence > there is no record of the PSTN number dialled. I guess most people want to > log the outgoing portion not the local call leg? Anyone know of a setting > that changes this? > > > Iain ___ 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
Subject: Re: [Asterisk-Users] Grandstream 100 sidetone
Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?
I was wondering if it is possible to have Asterisk push the media processing off to something with DSP's such as a gateway? That way, asterisk just has to handle the call setups and tear downs. Todd Wallace ___ 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 + BudgeTone (behind NAT)
I've been following up on my problem, which I previously described as: > I've concluded that the Netgear router (FVS318) performing the NAT is > corrupting the outgoing RTP packets. Traces confirmed that the BudgeTone > is sending them out with a UDP checksum of 0 but the next hop after the > Netgear router they are set to a non-zero value (an incorrect one). > Asterisk is never even seeing the packets because the kernel is > recognizing them as corrupt and dropping them, hence the recvfrom() > "Resource temporarily unavailable" errors in rtp.c. Here is Netgear's response: Original Message Subject: RE: Webform contact request [#] From:[EMAIL PROTECTED] Date:Fri, January 23, 2004 7:36 pm To: -- SIP VOIP phones do not work with netgear routers. The router will always set a value in the checksum. Regards Netgear Support [EMAIL PROTECTED] Please help us serve you better by clicking here mailto:[EMAIL PROTECTED] if you would like to provide any other valuable feedback. (Note: this feedback is not sent to an agent so you will not receive a reply.) -- Not exactly what I call stellar customer service! I realize this may not be the best solution -- actually, it's probably not even a good solution -- but has anyone experimented with using the Linux SO_NO_CHECK setsockopt() option? It looks like it could be used to ignore the checksums for the RTP packets. Owen ___ 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] 8 lines - best approach
Thanks Chris!! Running copper does not seem logical to me either. The last time I checked (Sept03), the cost of 8 lines in a T1 was almost double the cost of split lines. I have been considering NuFone, and have investigated it. We have also been looking for a decent IP based business phone, but I'll post a separate question for that :) I have three businesses all with the same problem. A decent T1 price would be the best, so I could centralize everything and only outsource the long distance side. I'm in Vancouver, BC Canada with the "Telus Inc" monopoly. What range of prices do most American telco's charge for a T1?? The price (in C$) I was quoted was $450/m plus $27/m per voice channel with zero features. Plus there was a $1200 setup fee. - Original Message - From: "Chris Albertson" <[EMAIL PROTECTED]> To: "Darren Martz" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 8:49 AM Subject: Fwd: [Asterisk-Users] 8 lines - best approach First get rid of those 8 analog lines. then you'l have two options: 1) Have the local phone company provide you with a T1 line that you can plug directly into the Digium card. After all it seems silly for ther phone company to split out the lines to 8 pairs runs 16 coppr wires only to have you re-combine them. 2) Get 8 DID numbers from a VOIP provider like NuFone or Iconnect and have all your incomming calls come in over your Internet link. Now yu've got zero hardawre, except for the PC. I suppose you would want local extensions... Depending on the numbr you might want a channel bank and anlog desk phones or go with all IP Phones --- Darren Martz <[EMAIL PROTECTED]> wrote: > From: "Darren Martz" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] 8 lines - best approach > Date: Fri, 23 Jan 2004 07:30:42 -0800 > > I have 8 lines coming into an existing PBX system and am looking for > a cost > effective way to replace the existing system with Asterisk. We need > some of > the features in Asterisk, including its ability to support remote > offices > (long distance savings). > > At first glance this appears to require a T100P card and a channel > bank, but > that seems rather expensive. My estimated price on that would be > roughly > $2600 for 8 lines given that system - perhaps my estimate is way > off > > Is there another way that is more cost effective? > > ___ > 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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] 8 lines - best approach
Thanks James!! The last quote I received from our local provider was had PRI at double the cost per line. The "access charge" was a major factor in the base cost. Alternative providers were more, even with using all 23 channels POTS was still cheaper than a T1. It doesn't make sense to me, but who ever said telco's made sense??? I will call again on Monday. - Original Message - From: "James H. Cloos Jr." <[EMAIL PROTECTED]> To: "Darren Martz" <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 11:21 AM Subject: Re: [Asterisk-Users] 8 lines - best approach > "Darren" == Darren Martz <[EMAIL PROTECTED]> writes: Darren> Is there another way that is more cost effective? Get a quote on an 8 trunk voice T1 and on an 8B+D pri from your telco -- and any clecs in the area. If it is competative you may not need the channel bank -JimC ___ 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] RFC3389 support issue with DG104S
Its silence supression. Turn that off and it will stop doing that. bkw On Fri, 23 Jan 2004, Zot O'Connor wrote: > I am getting (with older image): > > RFC3389 support incomplete. Turn off on client if possible > > > How do I turn that off for the DG104s? Or if I can't how do I tweak > asterisk? > > I see posts about ATA-186's having an audiomode, but the closet I came > to was inbanddtmf. I tried =0 and =1, no effect. > > Thanks! > > > -- > Zot O'Connor <[EMAIL PROTECTED]> > White Knight Hackers, Inc. > > ___ > 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] RFC3389 support issue with DG104S
I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <[EMAIL PROTECTED]> White Knight Hackers, Inc. ___ 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] Mediatrix 1204 sip experience?
See: the ATA section of: http://www.voip-info.org/wiki-VOIP+Phones and: http://www.voip-info.org/wiki-VoIP+Gateways for a list of what is available. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 12:34 PM Subject: Re: [Asterisk-Users] Mediatrix 1204 sip experience? > I'm not sure I understand your english here. I have two x100p's working just fine, > but I've got a couple more pstn lines I'd like to connect up. I probably could > put another one in the system, but I'd rather use a 4-port external gateway that > works well if such a thing exits at a reasonable price. (No, I don't want channel > banks and T1 cards for such a simple environment.) I'm just starting to do the > research on what is actually available. > > > Is it so hard to put X100P as a ethernet device? > > > > I have been trying FXO devices, but gets me luck. > > > > Kannaiyan > > > > - Original Message - > > > > > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip > > FXO > > > 4-port gateway? > > > > > > The archives tend to suggest the box is not very straight forward, and > > possibly > > > lacks some basic pstn interaction features. > > > > > > Thinking about trying one in place of a pair of x100p's (functioning fine > > now). > > > CallerId, etc, supported on this gateway? > > > > > > Rich > > > > > > > > > ___ > > > 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 > > ---End of Original Message- > > > ___ > 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] echo problems
New complile causing trouble... How do you enable MARK2 echo can with DAGGRESSIVE_ SUPPRESSION? looks like zconfig.h ..but I have quite a bit of echo on a bridgeded call and echocancelwhenbridged=no John -- This message was sent using Monroe-Woodbury's WebMail. http://webmail.mw.k12.ny.us/ This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ 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] Grandstream 100 sidetone
What firmware version do you have? On my phone the sidetone is very weak, I have to listen carfuly to hear it but there is no echo and the sound quality is good. It's to bad GS doesn't open source the software inside. I'd fix it in a minute by putting a volume setting on the admin web page. --- dkwok <[EMAIL PROTECTED]> wrote: > For people who are using GS 101, what do you think the sidetone > generated by the phone. > > I find mind a bit annoying. It has a delay and you notice it as an > echo. > The volume of the sidetone is also quite hight. I am distracted when > both caller and called party talking over each other occasssionally. > > The volume of the sidetone can be turned down using the volume button > > but it also control the volume of the voice call. As the sidetone is > louder than the conversation it is getting rather distracting. > > Can the sidetone be calibrated or adjusted? If not, how are people > coupling with it? > > -- > David Kwok > > Iaxtel/FWD # 17001813482 ext 1002 > > ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
NO it will log from a spool file if and only if you ref an extension and not an application. bkw On Fri, 23 Jan 2004, Kannaiyan Natesan wrote: > There is no CDR for the call from spool outgoing, > > You need to write a patch to solve the same. > > Kannaiyan > > - Original Message - > From: "Iain Stevenson" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, January 23, 2004 8:27 PM > Subject: [Asterisk-Users] Back to front logging for calls placed through > /var/spool/asterisk/outgoing? > > > > > > I've just noticed that if you start a call by writing a file to > > /var/spool/asterisk/outgoing the cdr created on termination logs the call > > placed to the local extension - not to the destination in the PSTN. Hence > > there is no record of the PSTN number dialled. I guess most people want > to > > log the outgoing portion not the local call leg? Anyone know of a setting > > that changes this? > > > > Iain > > ___ > > 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] G.723.1
If you don't have the licences for this codec, you can't playback files from *. If I'm not mistaken, * can be used to do codec passthrough between two endpoints, but you can't use any application to interact with *, like voicemail, directory, background or playback. Regards, Gus - Original Message - From: Cesar Rico To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 7:03 PM Subject: [Asterisk-Users] G.723.1 Hi all, I have a g.723.1 file and my voice devices support this codec, I need to playback this file in asterisk , I stored it in the directory /var/lib/asterisk/sounds/ but when I executte the command in the extension.conf (exten => 100,1,playback(file.g7323) the call hang up, my voice devices are configured with g723 codec, I read that * pass through this codec, so I don't know why this configuration don't work well, if anybody have some idea to respet let me know. I will appreciate you support Best regards Cesar Rico. ___ 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] Multiple voices on 64K channel (was) simple question...
On Thursday, January 22, 2004 9:55 PM, Jess Magnaye [SMTP:[EMAIL PROTECTED] wrote: > in telco world, there's like 64kbps per channel and voice can be > carried on a 16kbps channel. is it possible to configure asterisk to > make 4 extensions (ATAs example), to call out using single FXO port > at the same time? if that is possible, then is it also possible to > make t1-pri to be capable of transmitting 4x23ch simultaneous > calls..? In short you can not combine 4 calls on one FXO port and neither can the telcos. An FXO port, could not handle 4 channels of signaling since signaling is done either inband or with voltage changes such as polarity reversals. You would need other methods for multiple channels. On a 64K digital channel this is possible but only if both ends of the 64K channel have the capability. I don't believe any of the common PRI ISDN services could do it. As it stands now, asterisk could not do this. It would probably be better to explore trunking technologies such as low bandwidth codecs over IAX2. Don Pobanz ___ 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] Mediatrix 1204 sip experience?
Rich Adamson wrote: > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 > Sip FXO 4-port gateway? I have setup a Mediatrix 1204 with there SIP setup. They work! ??? Yes! and No. > The archives tend to suggest the box is not very straight forward, > and possibly lacks some basic pstn interaction features. They are very hard to configure. But on the other hand they have so many settings that they can almost be your PBX system. (Note the almost). > Thinking about trying one in place of a pair of x100p's (functioning > fine now). CallerId, etc, supported on this gateway? as your primary means of getting your lines in. This will slow the inbound calls. It will add 2 rings before it transfer the call to your system. > > Rich > > > ___ > 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] DG104S firmware has error?
I am installing a used DG104S I got it to ring from gnophone, but all I got was fast busies. so I upgraded based on Pavel's link: ftp://ftp.dlink.ru/pub/VoIP/DG-104S/Firmware/MGCPDG104.zip So I now have: PROM Version: 3.0B22-DRUNTIME Version: 3.0B44-D But when I pick the phone up I get: ggdbg>01604 DIM: 0 DSP ERROR: Reason= DIM ERROR: State Timeout 01604 DIM: 0:*, State Timeout Error (State = WAIT_RESTART_IND) 01604 DIM: 0:*, BRINGING DSP DOWN !! 01604 DSP 0 Failure. Error: -1 01604 CCU: 0, DSP Failure. State: 2 01604 xGCP: ERROR 0X105 in .\src\ggxgcpif.c at line 3761 (coding err) 01604 CCU: 0, Coding now (officially) free 01604 tcid 0. NO DSP RESOURCES 01604 xgcpimh: CA message rejected Is there another setting I have that is causing this? I did a factory reset. I then downgraded to dg104S_runtime_b35.bin But it would not keep the changes to the CA string, either via the web or console... Any ideas? Thanks! -- Zot O'Connor <[EMAIL PROTECTED]> White Knight Hackers, Inc. ___ 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] Latest cvs * compile error anyone?
Why is there still MYSQL stuff in there, I thought we had to remove that. On Fri, 23 Jan 2004, Kannaiyan Natesan wrote: > If you are not users from mysql database then you can disable in the > makefile. > > For this, > > USE_MYSQL_FRIENDS=1 > > change it to > > USE_MYSQL_FRIENDS=0 > > You won't get that error. > > Alternatively you can install mysqlclient library to compile it without > errors. > > Kannaiyan > > > - Original Message - > From: "SamW" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, January 23, 2004 7:03 PM > Subject: [Asterisk-Users] Latest cvs * compile error anyone? > > > > I downloaded asterisk and was trying to compile fresh, It end up in > > error, Any help appreciated. > > > > cvs checkout asterisk > > cd asterisk > > make clean > > make > > > > END UP with following error, (Previously I was able to compile without > > any errors. After a make clean stopped compiling.) > > > > gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o > > -lmysqlclient -lz > > /usr/bin/ld: cannot find -lmysqlclient > > collect2: ld returned 1 exit status > > make[1]: *** [chan_iax2.so] Error 1 > > > > > > - SamW > > > > > > ___ > > 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
[Asterisk-Users] Excternip and FWD
Hi I have updated from CVS about a week ago and got the externip working with FWD for outbound calls., but I'm having problems with inbound calls, I don't think they are even reaching the Asterisk box even though I have forwaorded 5060 and the rtp range specified, another thing I have notice, is that very ocasionally when I m using sip debug the internal IP is used for communication with FWD, Has anyone got inbound calls working with FWD and Externip through a Nat? Robb ___ 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] Mediatrix 1204 sip experience?
I'm not sure I understand your english here. I have two x100p's working just fine, but I've got a couple more pstn lines I'd like to connect up. I probably could put another one in the system, but I'd rather use a 4-port external gateway that works well if such a thing exits at a reasonable price. (No, I don't want channel banks and T1 cards for such a simple environment.) I'm just starting to do the research on what is actually available. > Is it so hard to put X100P as a ethernet device? > > I have been trying FXO devices, but gets me luck. > > Kannaiyan > > - Original Message - > > > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip > FXO > > 4-port gateway? > > > > The archives tend to suggest the box is not very straight forward, and > possibly > > lacks some basic pstn interaction features. > > > > Thinking about trying one in place of a pair of x100p's (functioning fine > now). > > CallerId, etc, supported on this gateway? > > > > Rich > > > > > > ___ > > 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 ---End of Original Message- ___ 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] Mediatrix 1204 sip experience?
UI for switch config allows you to generate scripts for setting if you need them. I found that to be useful. They can be easily configured from remote if you have the UI software. There are features for caller id, but I have not used them yet. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Friday, January 23, 2004 2:40 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] Mediatrix 1204 sip experience? Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ 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] MI2
It's NI-2 Yes it does. Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Welter Sent: Friday, January 23, 2004 12:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MI2 My CLEC just called and asked if we will support the "MI2" protocol on our proposed T1 circuit. I think this is for CallerID name. Will the T100P support this? Thanks, Mike ___ 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] G.723.1
Hi all, I have a g.723.1 file and my voice devices support this codec, I need to playback this file in asterisk , I stored it in the directory /var/lib/asterisk/sounds/ but when I executte the command in the extension.conf (exten => 100,1,playback(file.g7323) the call hang up, my voice devices are configured with g723 codec, I read that * pass through this codec, so I don’t know why this configuration don’t work well, if anybody have some idea to respet let me know. I will appreciate you support Best regards Cesar Rico. <>
[Asterisk-Users] RE: Latest cvs * compile error anyone?
I was getting that error today as well.. I just checked out a new CVS and it seems to be compiling now though. Try it again. ___ 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] Mediatrix 1204 sip experience?
Rich Adamson wrote: Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? A+ rating for me. What I don't like about 1204: I do not have any windows boxes. The SNMP code they provide to configure the box runs on windows. I had to borrow a lap top just to configure the box. I could not even read the MIB's off the CD on my Sun or linux box. What ever is on that CD really whacks out my Sun and Linux box when I try to read it. What I like about the 1204: After I got them running, I have never had to go back in and do anything to them. Every now and then I will peek at the syslog messages. I have had power outages, many * restarts/reloads and many linux reboots. The 1204's just keep running. I really like that. I has been so long since I configured them, I do not even remember how. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
There is no CDR for the call from spool outgoing, You need to write a patch to solve the same. Kannaiyan - Original Message - From: "Iain Stevenson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 8:27 PM Subject: [Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing? > > I've just noticed that if you start a call by writing a file to > /var/spool/asterisk/outgoing the cdr created on termination logs the call > placed to the local extension - not to the destination in the PSTN. Hence > there is no record of the PSTN number dialled. I guess most people want to > log the outgoing portion not the local call leg? Anyone know of a setting > that changes this? > > Iain > ___ > 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] Latest cvs * compile error anyone?
If you are not users from mysql database then you can disable in the makefile. For this, USE_MYSQL_FRIENDS=1 change it to USE_MYSQL_FRIENDS=0 You won't get that error. Alternatively you can install mysqlclient library to compile it without errors. Kannaiyan - Original Message - From: "SamW" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 7:03 PM Subject: [Asterisk-Users] Latest cvs * compile error anyone? > I downloaded asterisk and was trying to compile fresh, It end up in > error, Any help appreciated. > > cvs checkout asterisk > cd asterisk > make clean > make > > END UP with following error, (Previously I was able to compile without > any errors. After a make clean stopped compiling.) > > gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o > -lmysqlclient -lz > /usr/bin/ld: cannot find -lmysqlclient > collect2: ld returned 1 exit status > make[1]: *** [chan_iax2.so] Error 1 > > > - SamW > > > ___ > 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] MI2
Sorry, it's the "NI2" protocol. A previous poster said the "N2" protocol was supported by the T400P. Is "NI2" the same as "N2"? Does "NI2" mean "switchtype=national"? Thanks Michael Welter wrote: My CLEC just called and asked if we will support the "MI2" protocol on our proposed T1 circuit. I think this is for CallerID name. Will the T100P support this? Thanks, Mike ___ 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] Grandstream 100 sidetone
For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the volume of the voice call. As the sidetone is louder than the conversation it is getting rather distracting. Can the sidetone be calibrated or adjusted? If not, how are people coupling with it? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Troubles with the System Attendent Patch.
Title: Troubles with the System Attendent Patch.
Dear all,
I have spent some time tying to get the system attendant patch to work;
http://bugs.digium.com/bug_view_page.php?bug_id=214
I get no errors patching the system and the function runs, but I keep getting the following error;
queue: Nexus1, options: (null), url: (null), announce: (null), timeout: 0
-- Started music on hold, class 'default', on SIP/phone10-a3f0
-- Stopped music on hold on SIP/phone10-a3f0
Jan 23 15:19:04 WARNING[1226062640]: file.c:446 ast_openstream: File "queue-youarenext" ("You are now first in line.") does not exist in any format
Jan 23 15:19:04 WARNING[1226062640]: file.c:734 ast_streamfile: Unable to open "queue-youarenext" ("You are now first in line.") (format ULAW): No such file or directory
-- Told SIP/phone10-a3f0 in Nexus1 their queue position (which was 1)
Jan 23 15:19:04 WARNING[1226062640]: file.c:446 ast_openstream: File "queue-thankyou" ("Thank you for your patience.") does not exist in any format
Jan 23 15:19:04 WARNING[1226062640]: file.c:734 ast_streamfile: Unable to open "queue-thankyou" ("Thank you for your patience.") (format ULAW): No such file or directory
I have put the files in exits in var/lib/asterisk/sounds.
I'm running Redhat 9.0, Latest CVS, on a Dell 650. I have recompiled the Asterisk program several times and get to the same point.
Any help or suggestions would be most welcome.
Warm Regards
Shad Mortazavi
-
US Technical Manager
Nexus Management
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Go for inter-fone products. it can both support sip and h323. - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 2:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO > 4-port gateway? > > The archives tend to suggest the box is not very straight forward, and possibly > lacks some basic pstn interaction features. > > Thinking about trying one in place of a pair of x100p's (functioning fine now). > CallerId, etc, supported on this gateway? > > Rich > > > ___ > 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] Mediatrix 1204 sip experience?
Is it so hard to put X100P as a ethernet device? I have been trying FXO devices, but gets me luck. Kannaiyan - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 7:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO > 4-port gateway? > > The archives tend to suggest the box is not very straight forward, and possibly > lacks some basic pstn interaction features. > > Thinking about trying one in place of a pair of x100p's (functioning fine now). > CallerId, etc, supported on this gateway? > > Rich > > > ___ > 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] Mediatrix 1204 sip experience?
Works okay but user interface is a little like using RegEdit to program your router. In the version of software the one I have it lack some security features and I am unable to find any DMTF controls - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 2:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO > 4-port gateway? > > The archives tend to suggest the box is not very straight forward, and possibly > lacks some basic pstn interaction features. > > Thinking about trying one in place of a pair of x100p's (functioning fine now). > CallerId, etc, supported on this gateway? > > Rich > > > ___ > 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: AW: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
On Fri, 2004-01-23 at 00:33, Martin Bene wrote: > Hi Siggi/Jan, > The "Error Verifying Config Info" Message doesn't have anything to do with > the real problem. I also get that message, possibly because I don't keep a > device specific config file (SEP000D65707B78.cnf.xml) or > DISTINCTIVERINGLIST.XML on my tftp server. > > The real problem is keepalive timing: > > * the 7940 doesn't like the 5 second default timeout - set timeout to > >=10 seconds and you avoid the restart loop. That was it Thank you so much! Now things are working just fine. Now I just have to figure out why I'm not getting DTMF tones decoded *reliably* from my SIP provider. ...Jeff ___ 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] UK BT Interface with asterisk?
- Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 8:07 PM Subject: Re: [Asterisk-Users] UK BT Interface with asterisk? > Kannaiyan Natesan said: > > Do they offers, free evening and weekend calls? I get from BT. > > You can get a free 0870 number from http://www.speak2world.com but they > > charge for it. > > > > Kannaiyan > > > Don't think so but sometimes "free" isn't free. Depending on calling > patterns it might actually be lower cost to pay by the minute with a low > per minute rate. I talk for 5 to 7 hours with a speakerphone with BT. No extra charges. Completely FREE. Also with http://www.speak2world.com I get a free 0870 number with G711 codec. Perfectly clear voice. Kannaiyan ___ 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] 8 lines - best approach
Rich Adamson wrote: Two Voicetronix Openline 4 port FXO cards would do the trick. They run about $550 each. Their web site does not mention asterisk drivers. Is this card supported? The drivers are for Linux. Yes the card is supported. See vpb.conf. Any idea how it compares to a pair of external Mediatrix 1204 Sip FXO boxes? I have no tested Mediatrix. But with Multitech 8 ports, I do not see any difference from the FXO point of view. Jorge ___ 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] rc.local dont works
Hi ! thanks for the answer.. I use rh9... > I think with an interrupt problem, any startup will fail, may it be > manual or automatic during startup. but.. you think that there is a problem in the interrupts at all? i don´t understand. regards Miklos - Original Message - From: "Karsten Wemheuer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 5:07 PM Subject: Re: [Asterisk-Users] rc.local dont works > Hi > > listas iPfone wrote: > > Hi All > > > > I have a problem with initialization of asterisk using my rc.local > > file. when i call asterisk from the prompt it works well but don´t in > > the initialization... > > If it works when called directly and it doesn't work called during > startup. I would think, it is a problem with "path"-setting or with > access rights. > The init-scripts normaly have a relative short path. Maybee the > executable is not found. Or Your setup starts * as a non-root user, but > your manual startup uses root. > > What distribution do You use? > > > I have in my file that comands: > > > > touch /var/lock/subsys/local > > modprobe zaptel > > modprobe wcfxo > > safe_asterisk > > > > I read in somewere that it can be an interrup problem and i use the > > cat proc/interrupt to see what is happening > > I think with an interrupt problem, any startup will fail, may it be > manual or automatic during startup. > > HTH, > > Karsten > > > ___ > 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] 8 lines - best approach
It is supported. chan_vpb is the Voicetronix driver and I believe the asterisk file that deals with this is vpb.conf. Look in this month's mailing list archives for one user's (successful) experiences with this card. I have no idea about the comparison to Mediatrix. You do have the sip phones on the backend of this, right? John - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 1:33 PM Subject: Re: [Asterisk-Users] 8 lines - best approach > > > Two Voicetronix Openline 4 port FXO cards would do the trick. They run about $550 each. > > Their web site does not mention asterisk drivers. Is this card supported? > > Any idea how it compares to a pair of external Mediatrix 1204 Sip FXO boxes? > > > > ___ > 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] Debian Packages and Mirrors
On Fri, Jan 23, 2004 at 08:45:07AM -0800, Kostur, Andre wrote: > > v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not > on an i386) Ah, I didn't realize 0.7.1 was in unstable -- I run mostly testing here. > What do you have different in your packages? Nothing in particular as far as I know... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ 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] Latest cvs * compile error anyone?
I downloaded asterisk and was trying to compile fresh, It end up in error, Any help appreciated. cvs checkout asterisk cd asterisk make clean make END UP with following error, (Previously I was able to compile without any errors. After a make clean stopped compiling.) gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o -lmysqlclient -lz /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make[1]: *** [chan_iax2.so] Error 1 - SamW ___ 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] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain ___ 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] MI2
My CLEC just called and asked if we will support the "MI2" protocol on our proposed T1 circuit. I think this is for CallerID name. Will the T100P support this? Thanks, Mike ___ 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] UK BT Interface with asterisk?
Kannaiyan Natesan said: > Do they offers, free evening and weekend calls? I get from BT. > You can get a free 0870 number from http://www.speak2world.com but they > charge for it. > > Kannaiyan > Don't think so but sometimes "free" isn't free. Depending on calling patterns it might actually be lower cost to pay by the minute with a low per minute rate. ___ 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] Mediatrix 1204 sip experience?
Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ 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] R2 support
On Fri, 23 Jan 2004 15:49:31 -0300 "CW_ASN - Gus" <[EMAIL PROTECTED]> wrote: > < In Brasil, Telefonica offers ISDN, but it's a diferent comercial > < service (if you want voice and data in your E1), and it's more > < expensive. If you only want voice, the only choice is R2. > Very weird, in Argentina the cost is different only for international > calls in nx64; for national uses, the prices are the same for voice or > data. Here, the installation and all the contract are more expensive. sucks a lot. ___ 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] Debian Packages and Mirrors
Title: RE: [Asterisk-Users] Debian Packages and Mirrors hi everybody... http://www.backports.org has asterisk 0.7.1 for woody ;)) bye - Original Message - From: Kostur, Andre To: '[EMAIL PROTECTED]' Sent: Friday, January 23, 2004 5:45 PM Subject: RE: [Asterisk-Users] Debian Packages and Mirrors Note that there are also asterisk packages in the standard Debian repositories http://packages.debian.org/cgi-bin/search_packages.pl?keywords=asterisk&searchon=names&subword=1&case=insensitive&version=all&release=all v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not on an i386) The source for the zaptel interface is there too (package name: zaptel). Haven't looked for libpri... we don't have a PRI service...) What do you have different in your packages?
Re: [Asterisk-Users] Debian Packages and Mirrors
hi everybody... have you checked the asterisk backports from www.backports.org? I'm currently building my asterisk system and i think i will use these debs as I've successfully used alot of debs from backports.org in almost every production-server we have. don't know the quality of the asterisk packages from backports.org but I'm almost sure they are great ;)) bye thorsten - Original Message - From: "William Waites" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 5:08 PM Subject: [Asterisk-Users] Debian Packages and Mirrors > FYI and to whom it may concern, I have made Debian > packages of Asterisk et. al. You still need to build > a new kernel and the zaptel modules from source, but > Asterisk and libpri are manageable with dpkg. > > The debs as well as mirrors of the source distribution > are here: > > http://www.ntgos.com/Projects/Asterisk/Download > http://parc.styx.org/asterisk > > I would also like to mirror the CVS repository as > well as set up a cvsweb... > > -w > -- > /~\ The ASCII Ribbon Campaign > \ /No HTML/RTF in email > X No Word docs in email > / \ Respect for open standards > ___ > 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] 8 lines - best approach
Do you have to continue to use the existing handsets? You should look at replacing the existing phones with SIP phones. Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, January 23, 2004 8:04 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Fri, 2004-01-23 at 09:30, Darren Martz wrote: > I have 8 lines coming into an existing PBX system and am looking for a cost > effective way to replace the existing system with Asterisk. We need some of > the features in Asterisk, including its ability to support remote offices > (long distance savings). > > At first glance this appears to require a T100P card and a channel bank, but > that seems rather expensive. My estimated price on that would be roughly > $2600 for 8 lines given that system - perhaps my estimate is way off > > Is there another way that is more cost effective? That number sounds about right. It is likely that it will be less, but budgeting that much for hardware is a good start. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] 8 lines - best approach
> Two Voicetronix Openline 4 port FXO cards would do the trick. They run about $550 > each. Their web site does not mention asterisk drivers. Is this card supported? Any idea how it compares to a pair of external Mediatrix 1204 Sip FXO boxes? ___ 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] UK BT Interface with asterisk?
Do they offers, free evening and weekend calls? I get from BT. You can get a free 0870 number from http://www.speak2world.com but they charge for it. Kannaiyan - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 6:44 PM Subject: Re: [Asterisk-Users] UK BT Interface with asterisk? > Kannaiyan Natesan said: > > Have anyone tried to interface BT's Broadband Voice with asterisk? > > > > Kannaiyan > > ___ > > > No, and not sure of their rates but http://www.telappliant.com/ has good > rates, voice quality and is easy to interface to Asterisk. > Robert > ___ > 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] 8 lines - best approach
> "Darren" == Darren Martz <[EMAIL PROTECTED]> writes: Darren> Is there another way that is more cost effective? Get a quote on an 8 trunk voice T1 and on an 8B+D pri from your telco -- and any clecs in the area. If it is competative you may not need the channel bank -JimC ___ 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] rc.local dont works
Hi listas iPfone wrote: > Hi All > > I have a problem with initialization of asterisk using my rc.local > file. when i call asterisk from the prompt it works well but don´t in > the initialization... If it works when called directly and it doesn't work called during startup. I would think, it is a problem with "path"-setting or with access rights. The init-scripts normaly have a relative short path. Maybee the executable is not found. Or Your setup starts * as a non-root user, but your manual startup uses root. What distribution do You use? > I have in my file that comands: > > touch /var/lock/subsys/local > modprobe zaptel > modprobe wcfxo > safe_asterisk > > I read in somewere that it can be an interrup problem and i use the > cat proc/interrupt to see what is happening I think with an interrupt problem, any startup will fail, may it be manual or automatic during startup. HTH, Karsten ___ 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] R2 support
< In Brasil, Telefonica offers ISDN, but it's a diferent comercial < service (if you want voice and data in your E1), and it's more < expensive. If you only want voice, the only choice is R2. Very weird, in Argentina the cost is different only for international calls in nx64; for national uses, the prices are the same for voice or data. ___ 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] UK BT Interface with asterisk?
Kannaiyan Natesan said: > Have anyone tried to interface BT's Broadband Voice with asterisk? > > Kannaiyan > ___ > No, and not sure of their rates but http://www.telappliant.com/ has good rates, voice quality and is easy to interface to Asterisk. Robert ___ 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: Asterisk vs. Websphere Voice Response?
Thanks for all your responses on this, but on and off line... I got my questions answered, at least for now. E.A. (Ted) Thomas, CEO eThomasGroup.biz "Collaboration for Business" 502-802-6130 __ I am a collaboration consultant doing some research for a major client who has a problem. They have an existing IBM Direct Talk 2 IVR which they were going to upgrade to the current Websphere product, but choked on the price. My task is to find out the following: 1. The spec calls for a 24 analog line system with a fairly sophisticated response matrix using SQL into Oracle & text-to-speech (among other things). Is Asterisk in the same class of product as Websphere, or is it for a more straightforward "voicemail" office environment? 2. I assume Asterisk runs on PC class servers using PCI cards for incoming lines. Can the hardware side of a 24 line system be put together for less than $10K, or is it typically a lot more than that? 3. Can Asterisk be administered and/or programmed remotely? 4. Is Asterisk being used by any Fortune 500 companies in mission critical settings that you know of? Sorry for the general nature of the questions; although I use and promote open source products such as Linux, I am not that familiar with IVR servers. Thanks for any help you can give me... Sincerely, E.A. (Ted) Thomas, CEO eThomasGroup.biz "Collaboration for Business" 502-802-6130 ___ 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] Grandstream 101
On Thu, 22 Jan 2004, dkwok wrote: > Just got GS 101 phone and plugged into the network. > > Got ip setup however, the following problems arise: > > 1. when dialing an extension, I cannot further send any key tone to > Asterisk. > 2. there is no sound coming from the other end. > [gs] > canreinvite=no > dtmfmode=info To solve 1., use dtmfmode = rfc2833 or just leave it empty. > In the GS101 setting > rtp port = 5004 > sip port = 5060 > dtmf = sip info using "via RTP (RFC2833)" here works fine for me. > codec = pcmu > codec = pcma > > Any pointer of a sample of config file would be most appreciate. WRT the codecs, Setting all 6 choices in the grandstream web interface has helped for me, most of the time. One phone required several reset cycles before it would accept new settings, though. Another one only accepted the new settings after unplugging/replugging the power supply. This one also lost its settings during another power supply. I guess these "phones" are just a bit flakey WRT their settings... HTH, Siggi ___ 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] PSTN incoming - both SIP & H323 always arrive in default context :-?
Some of you may remember seeing my issue using SIP for incoming calls from the PSTN: http://voip-info.org/wiki-Asterisk+cisco+FXO i.e. all incoming calls arrive in the default 'bogon-calls' context. Well, I tried again using H.323 & get exactly the same result (both for chan_h323 & chan_oh323) i.e. all attempts to put a type=peer in sip.conf or a type=user in h323.conf for my host are ignored/bypassed. Is this a bug? Luckily for me, I can firewall off the H.323 port to all bar this one IP, so I now have a workable solution...until I want to extend the H.323 gateway to other devices... Anyone get host=x.x.x.x to be able to bypass the default contexts with either SIP or H.323? Cheers, Fran. ___ 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] UK BT Interface with asterisk?
> BT broadband voice uses ATA-186s configured as MGCP devices. I think asterisk supports MGCP. I want to configure MGCP with asterisk to connect to my BT Broadband Voice. Do you have any idea relating to that. Kannaiyan ___ 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] OT: Canada's Primus introduces SIP localservice
> Actually no. If you look at the model number of the Dlink box you will > notice that last letter M. This designates MGCP. Ok, that said - primus is pretty bad at technical support since they have just avoided any technical questions I have had so far and given me a canned sales response. I have asked if sip is available or anything else, and asked about migrating specific numbers. Lets agree for now it is MGCP and nothing else is available - are there any plans anyone has to build the necessary MGCP client to talk to this ? Is there any "soft" protocol converter out there which can talk to mgcp as a client and something else as a server on another signalling protocol ? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Colin > Anderson > Sent: Thursday, January 22, 2004 10:46 AM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] OT: Canada's Primus introduces SIP > localservice > > If you look at the specs on the Dlink box that Primus gives you, you > will > see that it is SIP. > > > > I am sure Primus has a SIP platform because we have played with it. We > managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 > hard phones. Their PC-Phone app is also a SIP soft phone. If you are > registering to sip.iprimus.net then it is definitely their SIP platyform > not MGCP. > > David > [EMAIL PROTECTED] 1/21/2004 6:39:34 AM >>> > > I'm not sure Primus uses SIP. I think it's MGCP. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of David Liu > Sent: Tuesday, January 20, 2004 9:16 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] OT: Canada's Primus introduces SIP > localservice > > Hey Colin, > > Do let me know if Primus' SIP service can work with Asterisk. We > tried > setting it up like how you would for iconnecthere However, we even > failed to register in the first place! (Of course password and > username > are correct). > > Anyone else on the list successfully used Primus' SIP with Asterisk? > > David > [EMAIL PROTECTED] 1/20/2004 12:25:50 PM >>> > Primus in Canada has launched a SIP-based service to replace your > business > and residential POTS lines with a VoIP version. It's called > TalkBroadband > and it looks killer: > > http://www.primus.ca/en/residential/talkbroadband/index.html > > Basic service for $20 Cdn a month!! > > Local number portability!! > > Cheapo Primus LD rates!! > > They don't care where geographically you plug it in!! > > When you sign up, they ship you this Dlink puppy for free: > > ftp://ftp10.dlink.com/pdfs/products/DVG-1120/DVG-1120_ds.pdf > > It has 2 FXS ports + ethernet + POTS backup port > > My order's in already, I'll be pleased to tell Telus where to put > their > "value pricing" once I get it installed. If anyone in Canada wants to > know > my experiences with it, email me off-list next month. > ___ > 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 > ___ > 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 > ___ 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 wierdness after upgrade from 0.7.1 to CVS
Just upgraded from 0.7.1 to the latest CVS version yesterday. This introduced a slew of warnings on startup. About 20 or 25 of the first, then 5 of the second: Jan 23 10:58:49 WARNING[8201]: chan_sip.c:446 __sip_xmit: sip_xmit of 0x80ef064 (len 461) to 0.0.0.0 returned -1: Invalid argument Jan 23 10:58:55 WARNING[8201]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) I updated from CVS today, and I'm still getting the problem. Is this a configuration problem on my end, or is it a bug? Thanx, Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. No purchase necessary. ___ 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] OT: Canada's Primus introduces SIP localservice
Actually no. If you look at the model number of the Dlink box you will notice that last letter M. This designates MGCP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, January 22, 2004 10:46 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] OT: Canada's Primus introduces SIP localservice If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> [EMAIL PROTECTED] 1/21/2004 6:39:34 AM >>> I'm not sure Primus uses SIP. I think it's MGCP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Liu Sent: Tuesday, January 20, 2004 9:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: Canada's Primus introduces SIP localservice Hey Colin, Do let me know if Primus' SIP service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David >>> [EMAIL PROTECTED] 1/20/2004 12:25:50 PM >>> Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign up, they ship you this Dlink puppy for free: ftp://ftp10.dlink.com/pdfs/products/DVG-1120/DVG-1120_ds.pdf It has 2 FXS ports + ethernet + POTS backup port My order's in already, I'll be pleased to tell Telus where to put their "value pricing" once I get it installed. If anyone in Canada wants to know my experiences with it, email me off-list next month. ___ 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 ___ 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] 8 lines - best approach
Title: RE: [Asterisk-Users] 8 lines - best approach Two Voicetronix Openline 4 port FXO cards would do the trick. They run about $550 each. John - Original Message - From: Kostur, Andre To: '[EMAIL PROTECTED]' Sent: Friday, January 23, 2004 10:40 AM Subject: RE: [Asterisk-Users] 8 lines - best approach One solution that we're investigating is using a gateway product instead of a channel bank. There's a couple to choose from...take a look at http://www.voip-info.org/wiki-VoIP+Gateways Sounds like you want one of the FXO devices. We haven't actually purchased ours yet, but we're looking into buying one of the Vegastream 50 Analog units. 10 incoming FXO ports. Saves us the cost of a proper channel bank, and a T1 card for the * server. I'm not recommending them specifically (just an example... you may be able to get better prices from other resellers), but Atacomm (http://www.atacomm.com) lists the price of a Vega 50 FXO at $2350, a Multitech MVP810 at $2999, AudioCodes MP108 FXO at $1429. (US $) > -Original Message- > From: Darren Martz [mailto:[EMAIL PROTECTED]] > Sent: Friday, January 23, 2004 7:31 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] 8 lines - best approach > > > I have 8 lines coming into an existing PBX system and am > looking for a cost > effective way to replace the existing system with Asterisk. > We need some of > the features in Asterisk, including its ability to support > remote offices > (long distance savings). > > At first glance this appears to require a T100P card and a > channel bank, but > that seems rather expensive. My estimated price on that would > be roughly > $2600 for 8 lines given that system - perhaps my estimate is > way off > > Is there another way that is more cost effective? > > ___ > 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] UK BT Interface with asterisk?
BT broadband voice uses ATA-186s configured as MGCP devices. Bear in mind that BT only allows you to make 1 hour of calls per day as part of their inclusive minutes, and ip-to-ip is not free and makes part of their inclusive minutes. If you're looking for a uk service then why not try www.voiptalk.org. :-) Tan telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: 23 January 2004 16:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] UK BT Interface with asterisk? Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan ___ 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] 8 lines - best approach
Title: RE: [Asterisk-Users] 8 lines - best approach Stay away from Audiocodes. No support. Won't even answer notes. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Kostur, AndreSent: Friday, January 23, 2004 8:41 AMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] 8 lines - best approach One solution that we're investigating is using a gateway product instead of a channel bank. There's a couple to choose from...take a look at http://www.voip-info.org/wiki-VoIP+Gateways Sounds like you want one of the FXO devices. We haven't actually purchased ours yet, but we're looking into buying one of the Vegastream 50 Analog units. 10 incoming FXO ports. Saves us the cost of a proper channel bank, and a T1 card for the * server. I'm not recommending them specifically (just an example... you may be able to get better prices from other resellers), but Atacomm (http://www.atacomm.com) lists the price of a Vega 50 FXO at $2350, a Multitech MVP810 at $2999, AudioCodes MP108 FXO at $1429. (US $) > -Original Message- > From: Darren Martz [mailto:[EMAIL PROTECTED]] > Sent: Friday, January 23, 2004 7:31 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] 8 lines - best approach > > > I have 8 lines coming into an existing PBX system and am > looking for a cost > effective way to replace the existing system with Asterisk. > We need some of > the features in Asterisk, including its ability to support > remote offices > (long distance savings). > > At first glance this appears to require a T100P card and a > channel bank, but > that seems rather expensive. My estimated price on that would > be roughly > $2600 for 8 lines given that system - perhaps my estimate is > way off > > Is there another way that is more cost effective? > > ___ > 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] 8 lines - best approach
I have a similar setup currently with 7 incomming lines. We purchased an Adit 600 CB from Suntel data with 1 8 port fxo and 1 8 port fxs for $800. T100P for about $500. Cost is of course relative to the money you will save on line charges :-). -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting "Kostur, Andre" <[EMAIL PROTECTED]>: > One solution that we're investigating is using a gateway product instead of > a channel bank. > > There's a couple to choose from...take a look at > http://www.voip-info.org/wiki-VoIP+Gateways > > Sounds like you want one of the FXO devices. We haven't actually purchased > ours yet, but we're looking into buying one of the Vegastream 50 Analog > units. 10 incoming FXO ports. Saves us the cost of a proper channel bank, > and a T1 card for the * server. > > I'm not recommending them specifically (just an example... you may be able > to get better prices from other resellers), but Atacomm > (http://www.atacomm.com) lists the price of a Vega 50 FXO at $2350, a > Multitech MVP810 at $2999, AudioCodes MP108 FXO at $1429. (US $) > > > -Original Message- > > From: Darren Martz [mailto:[EMAIL PROTECTED] > > Sent: Friday, January 23, 2004 7:31 AM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] 8 lines - best approach > > > > > > I have 8 lines coming into an existing PBX system and am > > looking for a cost > > effective way to replace the existing system with Asterisk. > > We need some of > > the features in Asterisk, including its ability to support > > remote offices > > (long distance savings). > > > > At first glance this appears to require a T100P card and a > > channel bank, but > > that seems rather expensive. My estimated price on that would > > be roughly > > $2600 for 8 lines given that system - perhaps my estimate is > > way off > > > > Is there another way that is more cost effective? > > > > ___ > > 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 > > > Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] MGCP w/8x8 DTA-310 and as5300 pstn gateway
This one does mgcp... It's been used in conjunction with a hosted pbx system called Centile that 8x8 now owns. If there's a firmware image anyone knows of to make these do sip, I'd rather do that. But for now, mgcp help is what I need. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, January 23, 2004 9:08 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway The Packet8 8x8 DTA-310 that I have ran SIP when I was using it. ___ 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] rc.local dont works
Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don´t in the initialization... I have in my file that comands: touch /var/lock/subsys/localmodprobe zaptelmodprobe wcfxosafe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody can tell me if this is correct? The usb-ohci and usb-ohci drivers are to be sharing the same interrupt as the wcfxo? CPU0 0: 220155 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 68 XT-PIC eth1 8: 1 XT-PIC rtc 9: 2167768 XT-PIC usb-ohci, usb-ohci, wcfxo 10: 7320 XT-PIC eth0 12: 22 XT-PIC PS/2 Mouse 14: 5092 XT-PIC ide0NMI: 0ERR: 0 Thanks for any help Miklos
[Asterisk-Users] UK BT Interface with asterisk?
Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan ___ 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: SIP register/auth with Grandstream BudgeTone-100
Key Aavoja wrote: Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to register in asterisk Yes if Im trying to call somewhere. Registers before any calls are made. Probably your extension name and registration data don't match. Here is my SIP config and a list of the GS phone settings: [exten106] type=friend context=administrator callerid=<829-3289 106> username=sbesch host=dynamic dtmfmode=info ;or inband if you prefer secret=yourpassword qualify=5000 mailbox=106 canreinvite=no ;As long as the phones are NAT'ed The caller ID only means something to our internal extensions, since the phone company will not let me set callerid data. I don't think that the username is needed. I use it because it shows up in the CLI response to SIP SHOW PEERS and helps me identify the phone. The important bits are that the extension name (the part in "[]") and the secret match the data in the phone setup: Sip User ID: exten106 Authenticate ID: exten106 Aithentication Password: yourpassword Sip Registration: Yes Send DTMF: via SIP Info Don't make the mistake of thinking that the username entry in SIP.conf has anything to do with the Authenticate ID. IT doesn't. The only thing that works is to set the User ID and the Authenticate ID to the same thing. You may find that the GS phones will dissappear after a while if you use dynamic registration. Alas, this is a bug in the GS firmware(1.0.3.81). I don't know if it has been fixed in later releases. I am not willing to update my phones until the firmware gets much more stable - they are all working and my philosophy is that if it ain't broke, don't fix it - so I haven't been able to test this. If your phones have fixed addresses, you might as well specify the IP addresses in SIP.conf and preemptively avoid the problem of the GS registrations dissappearing. ___ 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] Debian Packages and Mirrors
Title: RE: [Asterisk-Users] Debian Packages and Mirrors Note that there are also asterisk packages in the standard Debian repositories http://packages.debian.org/cgi-bin/search_packages.pl?keywords=asterisk&searchon=names&subword=1&case=insensitive&version=all&release=all v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not on an i386) The source for the zaptel interface is there too (package name: zaptel). Haven't looked for libpri... we don't have a PRI service...) What do you have different in your packages? > -Original Message- > From: William Waites [mailto:[EMAIL PROTECTED]] > Sent: Friday, January 23, 2004 8:09 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Debian Packages and Mirrors > > > FYI and to whom it may concern, I have made Debian > packages of Asterisk et. al. You still need to build > a new kernel and the zaptel modules from source, but > Asterisk and libpri are manageable with dpkg. > > The debs as well as mirrors of the source distribution > are here: > > http://www.ntgos.com/Projects/Asterisk/Download > http://parc.styx.org/asterisk > > I would also like to mirror the CVS repository as > well as set up a cvsweb... > > -w > -- > /~\ The ASCII Ribbon Campaign > \ / No HTML/RTF in email > X No Word docs in email > / \ Respect for open standards
RE: [Asterisk-Users] 8 lines - best approach
Title: RE: [Asterisk-Users] 8 lines - best approach One solution that we're investigating is using a gateway product instead of a channel bank. There's a couple to choose from...take a look at http://www.voip-info.org/wiki-VoIP+Gateways Sounds like you want one of the FXO devices. We haven't actually purchased ours yet, but we're looking into buying one of the Vegastream 50 Analog units. 10 incoming FXO ports. Saves us the cost of a proper channel bank, and a T1 card for the * server. I'm not recommending them specifically (just an example... you may be able to get better prices from other resellers), but Atacomm (http://www.atacomm.com) lists the price of a Vega 50 FXO at $2350, a Multitech MVP810 at $2999, AudioCodes MP108 FXO at $1429. (US $) > -Original Message- > From: Darren Martz [mailto:[EMAIL PROTECTED]] > Sent: Friday, January 23, 2004 7:31 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] 8 lines - best approach > > > I have 8 lines coming into an existing PBX system and am > looking for a cost > effective way to replace the existing system with Asterisk. > We need some of > the features in Asterisk, including its ability to support > remote offices > (long distance savings). > > At first glance this appears to require a T100P card and a > channel bank, but > that seems rather expensive. My estimated price on that would > be roughly > $2600 for 8 lines given that system - perhaps my estimate is > way off > > Is there another way that is more cost effective? > > ___ > 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] 8 lines - best approach
On Fri, 2004-01-23 at 09:30, Darren Martz wrote: > I have 8 lines coming into an existing PBX system and am looking for a cost > effective way to replace the existing system with Asterisk. We need some of > the features in Asterisk, including its ability to support remote offices > (long distance savings). > > At first glance this appears to require a T100P card and a channel bank, but > that seems rather expensive. My estimated price on that would be roughly > $2600 for 8 lines given that system - perhaps my estimate is way off > > Is there another way that is more cost effective? That number sounds about right. It is likely that it will be less, but budgeting that much for hardware is a good start. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] MGCP w/8x8 DTA-310 and as5300 pstn gateway
The Packet8 8x8 DTA-310 that I have ran SIP when I was using it. On Thu, 2004-01-22 at 23:11, Jeremy Jones wrote: > Hello folks, > > I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no > dialtone & can't get it to ring. My mgcp.conf says: > > ; > ; MGCP Configuration for Asterisk > ; > [general] > port = 2427 > bindaddr = 0.0.0.0 > > [172.16.2.25] > host = 172.16.2.25 > context = default > line => aaln/1 > > And here's the interesting bits of extensions.conf: > > [globals] > ... > TRUNK=H323/[EMAIL PROTECTED] > ... > > [default] > exten => 2084728803,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) > > And, finally, the h323.conf: > > > ; The NuFone Network's > ; Open H.323 driver configuration > ; > [general] > port = 1720 > bindaddr = 0.0.0.0 > tos=lowdelay > allow=all ; turns on all installed codecs > disallow=g723.1; Hm... Proprietary, don't use it... > allow=gsm ; Always allow GSM, it's cool :) > gatekeeper = 10.0.0.202 > AllowGKRouted = yes > context=default > > [2084728803] > type=h323 > e164=2084728803 > context=default > > Now, including the demo context in default w/out the mgcp extension > worked just dandy & impressed everyone. I know my box is registering > with the gatekeeper just fine & the as5300 is getting calles to > 2084728803 number to the asterisk box. > > First: in my [globals] section, is that the right way to use the h323 > channel as a trunk? I couldn't find much relating to this out there. > > Second, what's up with my 8x8 gear? Anyone got it to work? I can > attach my "mgcp debug" output if someone likes. Here's a briefer bit > from "asterisk -c" when picking up the handset on the phone > connected to the dta-310: > > *CLI> Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: > Removing messa > ge from aaln/[EMAIL PROTECTED] tansaction 1 > -- Resetting interface aaln/[EMAIL PROTECTED] > -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' > -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down > Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 2 > Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 3 > Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 4 > -- Resetting interface aaln/[EMAIL PROTECTED] > -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' > -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 5 > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 6 > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 7 > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 8 > -- Resetting interface aaln/[EMAIL PROTECTED] > -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' > -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 9 > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 10 > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 11 > Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 12 > -- Resetting interface aaln/[EMAIL PROTECTED] > -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' > -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down > Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 13 > Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 14 > Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 15 > Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 16 > -- Resetting interface aaln/[EMAIL PROTECTED] > -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' > -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down > -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' > Jan 22 22:07:43 WARNING[213006]: chan_mgcp.c:2126 handle_hd_hf: Off > hook, but al > reaedy have owner on aaln/[EMAIL PROTECTED] > Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing > message fro > m aaln/[EMAIL PROTECTED] tansaction 17 > Jan 22 22:07:43 N
[Asterisk-Users] Debian Packages and Mirrors
FYI and to whom it may concern, I have made Debian packages of Asterisk et. al. You still need to build a new kernel and the zaptel modules from source, but Asterisk and libpri are manageable with dpkg. The debs as well as mirrors of the source distribution are here: http://www.ntgos.com/Projects/Asterisk/Download http://parc.styx.org/asterisk I would also like to mirror the CVS repository as well as set up a cvsweb... -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ 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] compiling * pipe error
Building * on a machine with a minimal install of Mandrake, worked fine on non minimal install now I get this: bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe If anyone can help me figure out what package I might have missed out when installing mandrake, it would be great. It is possible that the security system is causing a problem, but am not sure here either, any help is greatly appreciated Thanks Chris. ___ 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] Toll-Free Gateway Beta Test: freenum.org
John Todd said: United States:* +1-800-... +1-888-... +1-877-... +1-866-... via: Telesthetic/Local Exchange Carriers of Michigan JOhn, Good idea on leaving the code in. I'll do that. Since IAXtel has 8xx dialing into the USA would it be possible to have that one in freenum.org as the returned gateway fro the USA toll free? That way, IAX will get some testing too. Just a thought. Robert Two comments on that: 1) I think the same people that provide the actual gateways to IAXTel are the ones that offered their services for freenum.org (Telesthetic/Local Exchange Carriers of Michigan) 2) The +1-8xx numbers offered in freenum.org are now available via IAX2, thanks to the kind efforts of the gateway provider. Here's the issue: Asterisk does not have the ability to request specific NAPTR replies. Since I have two different technologies (SIP and IAX2) in the list of records for (as an example) 1-800, there are two possible replies: a SIP NAPTR and an IAX2 NAPTR. Now, the good news is that the standard DNS resolver on most UNIX distributions will rotate the two answers, so every time your particular computer asks the question from your local resolver you should get a different response than the previous request, or at least you will receive answers in a quasi-random fashion. So, in the best possible world, every other call you make will be IAX2. You could write some really ugly looping routines in Asterisk to keep requesting NAPTRs from EnumLookup until an IAX2 reply was given... . SO: Asterisk needs to have some extensions written to the EnumLookup routines that allow the user optionally to specify the technology requested. Where was that guy on the IRC channel that was looking for Asterisk projects... JT ___ 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] Snom 200 phones not working.
Hi I sugest you to make a reset and switch off the phone before upgrade. It solved many problems for me. Miklos - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 11:32 AM Subject: Re: [Asterisk-Users] Snom 200 phones not working. > Ariel, > > > I have 2 Snom 200 and would like to get them to work properly with > > Asterisk. With the Firmware 2.02t I am able to use the phone. But only > > one line configured. With there newer firmware 2.03o it will not allow > > me to make calls. But I can get calls on the unit. Again the 2nd line > > is not able to be registered. Is this an issue with Asterisk or Snom? > > > > I could use some example configuration files. I have followed the Snom > > FAQ step by step. But it's still not working. > > I just upgraded my 200 to v2.03o and its working fine with two extns > defined. I happen to be using * CVS-12/04/03-14:24:40 on the same wire > (no nat, etc). > > My sip.conf entries look like: > [3007] > type=friend > host=dynamic > username=3007 > secret=mypassword > context=from-sip > > [3008] > type=friend > host=dynamic > username=3008 > secret=mypassword > context=from-sip > > Using your web browser to config the phone, verify: > Settings/SIP/Lines > Account = 3007 (to match above sip.conf def) > Registrar = ip address of asterisk box > Action = proxy > Account = 3008 (to match above sip.conf def) > Registrar = ip address of asterisk box > Action = proxy > Settings/SIP/Stack > Outbound Proxy, Registrar is outbound proxy = yes > Settings/SIP/Authentication > Line 1, Realm = asterisk, Username = 3007, Pasword = mypassword > Line 2, Realm = asterisk, Username = 3008, Pasword = mypassword > Settings/Key Mapping > P1 = Line, Number = 3007 > P2 = Line, Number = 3008 > > After ensuring your phone settings actually match the sip.conf settings > and that you've properly selected "Save" after changing each of the above > entries in the phone, then reboot the phone. If the phone prompts you to > download another firmware image, simply press ESC. (Seems some config > changes don't take effect until after a phone reboot.) > > The above config has been working fine with the last several (estimate > about 10) firmware versions, however the "user" interaction with several of > the keys are rather non-intuitive (or even backwards) for US users. > > For example, if you answer an incoming call on Line 1 (x3007 above) and place > that call on hold using the "Hold" key, then select Line 2 (x3008) to do a > consultive call to a different extn, you have to press the ESC key to hang > up that second consultive call. If instead of pressing the ESC key you > simply press Line 1 to return to the original call, Line 2 is automatically > put on hold (instead of dropping the line as it does in the US). If you're > not paying attention to the LEDs, you've now tied up the second line/extn > until such time as you muck around to release it. > > If that second (consultive) call happens to be to a pstn user and your > Central Office supports calling-party line supervision, you've probably > tied up that person's telephone line as well. (Email comments to snom > resulted in push-back, suggesting the ESC key is the proper way to drop > that second line. I'd guess US users (not techie's) will object to using > the phone in any form of production telephony.) > > I've not tried the 200 with the later CVS versions, so don't have a clue as > to what you're milage may be. > > Rich > > > ___ > 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] 8 lines - best approach
I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? ___ 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] SIP Absolute Timeout
I think this has been fixed since 0.5.0 their was a problem with timeout's and native bridges. Might wanna update. bkw On Fri, 23 Jan 2004, Wes Marderness wrote: > Hi All, > > I've been having a hard time getting the AbsoluteTimeout function to work. > Is this Function working in for SIP? I've search all the messages in the > news letters and tried what was suggested and still have not gotten it to > work. Below is a portion of my extensions.conf. I've also been running these > test on ver 0.5.0 > > exten => _X.,1,Absolutetimeout(20) > exten => _X.,2,dial(SIP/[EMAIL PROTECTED]) > > exten => T,1,BackGround(tt-weasels) > exten => T,2,Hangup() > > Thanks ahead of time for any help / suggestions. > Wes Marderness > > ___ > 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] New Asterisk article on O'Reilly's onlamp.com
Here's the follow-up article to the first article I published on Asterisk: http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html?page=last This one covers getting Zap hardware installed, and also covers integrating an IPCSP (IP Communications Service Provider - aka: long distance via SIP.) Sorry for the delay - the article was submitted about five months ago and just finally surfaced on their site. JT ___ 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] SIP Absolute Timeout
I use it in that way, it works very well: exten => s,4,AbsoluteTimeout,600 miklos - Original Message - From: "Wes Marderness" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 12:33 PM Subject: [Asterisk-Users] SIP Absolute Timeout > Hi All, > > I've been having a hard time getting the AbsoluteTimeout function to work. > Is this Function working in for SIP? I've search all the messages in the > news letters and tried what was suggested and still have not gotten it to > work. Below is a portion of my extensions.conf. I've also been running these > test on ver 0.5.0 > > exten => _X.,1,Absolutetimeout(20) > exten => _X.,2,dial(SIP/[EMAIL PROTECTED]) > > exten => T,1,BackGround(tt-weasels) > exten => T,2,Hangup() > > Thanks ahead of time for any help / suggestions. > Wes Marderness > > ___ > 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] Snom 200 phones not working.
Ariel, > I have 2 Snom 200 and would like to get them to work properly with > Asterisk. With the Firmware 2.02t I am able to use the phone. But only > one line configured. With there newer firmware 2.03o it will not allow > me to make calls. But I can get calls on the unit. Again the 2nd line > is not able to be registered. Is this an issue with Asterisk or Snom? > > I could use some example configuration files. I have followed the Snom > FAQ step by step. But it's still not working. I just upgraded my 200 to v2.03o and its working fine with two extns defined. I happen to be using * CVS-12/04/03-14:24:40 on the same wire (no nat, etc). My sip.conf entries look like: [3007] type=friend host=dynamic username=3007 secret=mypassword context=from-sip [3008] type=friend host=dynamic username=3008 secret=mypassword context=from-sip Using your web browser to config the phone, verify: Settings/SIP/Lines Account = 3007 (to match above sip.conf def) Registrar = ip address of asterisk box Action = proxy Account = 3008 (to match above sip.conf def) Registrar = ip address of asterisk box Action = proxy Settings/SIP/Stack Outbound Proxy, Registrar is outbound proxy = yes Settings/SIP/Authentication Line 1, Realm = asterisk, Username = 3007, Pasword = mypassword Line 2, Realm = asterisk, Username = 3008, Pasword = mypassword Settings/Key Mapping P1 = Line, Number = 3007 P2 = Line, Number = 3008 After ensuring your phone settings actually match the sip.conf settings and that you've properly selected "Save" after changing each of the above entries in the phone, then reboot the phone. If the phone prompts you to download another firmware image, simply press ESC. (Seems some config changes don't take effect until after a phone reboot.) The above config has been working fine with the last several (estimate about 10) firmware versions, however the "user" interaction with several of the keys are rather non-intuitive (or even backwards) for US users. For example, if you answer an incoming call on Line 1 (x3007 above) and place that call on hold using the "Hold" key, then select Line 2 (x3008) to do a consultive call to a different extn, you have to press the ESC key to hang up that second consultive call. If instead of pressing the ESC key you simply press Line 1 to return to the original call, Line 2 is automatically put on hold (instead of dropping the line as it does in the US). If you're not paying attention to the LEDs, you've now tied up the second line/extn until such time as you muck around to release it. If that second (consultive) call happens to be to a pstn user and your Central Office supports calling-party line supervision, you've probably tied up that person's telephone line as well. (Email comments to snom resulted in push-back, suggesting the ESC key is the proper way to drop that second line. I'd guess US users (not techie's) will object to using the phone in any form of production telephony.) I've not tried the 200 with the later CVS versions, so don't have a clue as to what you're milage may be. Rich ___ 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] chan h323 Compile problem
Mike Bentley wrote: Hi can anyone help me with this g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I../../include -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp In file included from /usr/src/pwlib/include/ptlib.h:169, from ast_h323.h:30, from ast_h323.cpp:29: /usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: parse error before ` protected' Read the README in asterisk/channels/h323 Jeremy McNamara ___ 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] SIP register/auth with Grandstream BudgeTone-100
Key, I've been playing with the Grandstreams for some weeks; one good way to see the registration messages it to monitor the network with Ethereal. (packet sniffer). You'll see the SIP messages coming and going, with complete decoding. This works pretty much as predicted when using VOCAL. (another SIP server) That said, I recently asked about registration with Asterisk from the Grandstreams, and received a reply that if the phones have a fixed IP address they do not "register" as they are already accounted for in the sip.conf by the user name/number assigned to the phone, and indeed the phones work fine (at least inside the firewall). Other soureces have described Asterisk as having "limited or not fully implemented SIP support"... but I have never been able to determine to what extent SIP support is lacking. I was able to get the Grandstreams running only recently with Asterisk... My sip.conf for the Grandstreams has several differences from yours: For a grandstream phone with an assigned number of 1000 and a fixed IP address of 192.168.0.160 I have the following: [1000] type=friend username=1000 host=dynamic reinvite=no canreinvite=no qualify=300 callerid="Larry's Desk" <1000> mailbox=1000 nat=no dtmfmode=info disallow=all allow=ulaw allow=mlaw See the message I sent to Tom Scott on 1/21 for a little more information. I would very much like to put together a set of instructions for the Grandstream phones+asterisk that could expand on the scattered information found on the wiki/web/draft manual and this list. And let us know what you find! -- L ___ 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] Snom 200 phones not working.
Geert Nijpels wrote: > Ariel Batista wrote: >> > I had the same problem, so I emailed SNOM. After a quick and clear > reaction from SNOM, the following turns out: > > I have an SRV record set for Asterisk using both TCP and UDP, because > I was first experimenting with SER and that SIP proxy DOES support > TCP. Asterisk does not. So in firmware 2.02t the phone tried udp > automatically, and in firmware 2.03o the phone tried to use tcp, which > will not work in Asterisk. Removing the TCP SRV entry solved my > problem Maybe this will solve your problem. I found my main problem. It was the way I was setting it up. I was putting the servers IP address in the authentication area. This you have to put asterisk instead. And the IP address goes in the line area. > However, I still have other problems with the SNOM phones: > - All sound stops working sometimes (also ringtone) > - Speech sometimes not working (not sure if it is RTP problem or SNOM > firmware problem) > - Sometimes the phone returns BUSY when not busy in firmware 2.02t. > Resetting the phone or adding a new SIP line solves this. In firmware > 2.03o, this is different. The phone does not respond and Asterisk > gives the error: phone CIRCUIT BUSY. I have seen other posts here > from people having the same problems. I have upgraded to the 2.03o and I have both phones working just fine. I had the choppy sound and found out it was a network cable that was bad. I have it working with great sound now on gsm which before it would only work with alaw. ulaw seems to still have some what I call hickcups. You hear it sound but every now and then you hear a skip. > Other people having the same problems? It makes my case a bit more > clear at SNOM when I can point them to a thread with a lot of people > having the same problems:-) > > I am emailing with SNOM about these issues. > > Kind regards, > > Geert Nijpels > > ___ > 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 Absolute Timeout
Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten => _X.,1,Absolutetimeout(20) exten => _X.,2,dial(SIP/[EMAIL PROTECTED]) exten => T,1,BackGround(tt-weasels) exten => T,2,Hangup() Thanks ahead of time for any help / suggestions. Wes Marderness ___ 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] Nufone not taking GSM CALLS
[EMAIL PROTECTED] wrote: Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1 That error means asterisk cannot transcode to GSM Make sure you have not mistakenly noload'ed codec_gsm.so or havent' disallowed it in your iax.conf. Jeremy McNamara ___ 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] chan h323 Compile problem
Do not compile openh323 and pwlib from cvs.
Use the versions described in the README of chan_h323 so (in
channels/h323 dir).
Good luck!
Doichin Dokov
Mike Bentley wrote:
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC
-Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DPTRACING -DP_USE_PRAGMA -I../../include
-I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include
-I/usr/src/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from /usr/src/pwlib/include/ptlib.h:169,
from ast_h323.h:30,
from ast_h323.cpp:29:
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: parse error
before `
protected'
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: syntax error
before `*'
token
In file included from /usr/src/pwlib/include/ptlib.h:181,
from ast_h323.h:30,
from ast_h323.cpp:29:
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:53: parse error
before `public
'
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:55: destructors must
be member
functions
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:57: parse error before `
protected'
In file included from /usr/src/pwlib/include/ptlib.h:187,
from ast_h323.h:30,
from ast_h323.cpp:29:
/usr/src/pwlib/include/ptlib/args.h:121: parse error before `{' token
/usr/src/pwlib/include/ptlib/args.h:147: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:156: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:165: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:175: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:190: `ostream' was not declared in
this
scope
/usr/src/pwlib/include/ptlib/args.h:191: `strm' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:191: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:191: variable or field `PrintOn'
declared
void
/usr/src/pwlib/include/ptlib/args.h:197: `istream' was not declared in
this
scope
/usr/src/pwlib/include/ptlib/args.h:198: `strm' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:198: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:198: variable or field `ReadFrom'
declared
void
/usr/src/pwlib/include/ptlib/args.h:206: parse error before `&' token
/usr/src/pwlib/include/ptlib/args.h:215: parse error before `&' token
/usr/src/pwlib/include/ptlib/args.h:246: virtual outside class
declaration
open h323 and pwlib from cvs complied fine.
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RE: [Asterisk-Users] Polycom Reboot Script - Please wiki-size me
It's been added to the wiki:
http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script
MATT---
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Friday, January 23, 2004 12:38 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom Reboot Script - Please wiki-size me
With my thanks to Brian West and his offering in the thread,
"Subject: Re: [Asterisk-Users] Remote reload Cisco 7960"
I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones
PolyReboot.pl takes an IP address as a single argument and reboots the
phone.
You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all
lower case. Further,
you need to use ftp for your remote cfg loading, because with ftp
synchronization, the phones
check the ctimes of the cfg files against what's in their cache when
receiving a check-sync in
order to determine if rebooting is necessary.
I tested this script. It works on my Polycoms. I have the latest bootrom
and the
SIP 1.1.0 software version.
I'm not much of a perl person, and all of this code is heavily borrowed,
(Thanks again, Brian), so
someone else can prettify it as they see fit.
#!/usr/bin/perl
#
# PolyReboot.pl
#
# Reboots a Polycom 500 or 600 phone
#
use Net::Ping;
use Socket;
$polycompath = '/home/poly/';# Where you keep your config files
$arp = '/sbin/arp'; # Location of arp command
$sipserver = '192.168.XXX.XXX'; # IP of asterisk server
$phone = shift;
checkphone("$phone");
touch( arp2config("$phone") );
reboot_sip_phone( "$phone", "$sipserver", "Reboot" );
sub checkphone { # Checks for existence of phone, makes sure
# it's in arp table
$activephone = shift;
# ARP table needs our phone
print "Checking ARP table.\n";
$p = Net::Ping->new("icmp");
if ( $p->ping( $activephone, 2 ) ) {
print "$activephone is ";
print "reachable.\n";
}
else { die "Polycom at ", $activephone, " is not reachable!"; }
sleep(1);
$p->close();
}
sub arp2config {# Gets mac address from arp table, converts
# to a polycom config filename, makes sure
# the config file exists
$arpip = shift;
open( ARP, "$arp -an|" ) || die "Couldn't open arp table: $!\n";
print "checking for polycom config name...", "\n";
while () {
chomp;
$addr = $_;
$ip = $_;
$addr =~ s/.*
([\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+).*/$1/;
$addr =~ s/://g;
$addr = lc($addr) . '.cfg';
$ip =~ s/.*?(\d+\.\d+\.\d+\.\d+).*/$1/;
if ( $ip eq $arpip ) {
last;
}
}
$polycomconfig = "$polycompath" . "$addr";
unless ( -e "$polycomconfig" ) {
print "sorry, polycom config file ", "$polycomconfig",
" is not found.\n\n";
exit;
}
return $polycomconfig;
}
}
$polycomconfig = "$polycompath" . "$addr";
unless ( -e "$polycomconfig" ) {
print "sorry, polycom config file ", "$polycomconfig",
" is not found.\n\n";
exit;
}
return $polycomconfig;
}
sub touch {# We need to touch the config file or the phone
# won't reboot - it depends on time synchronization
print "touching config file ", $polycomconfig, "\n";
my $now = time;
local (*TMP);
foreach my $file (@_) {
utime( $now, $now, $file )
|| open( TMP, ">>$file" )
|| die ("$0: Couldn't touch file: $!\n");
}
}
sub reboot_sip_phone {# Send the phone a check-sync to reboot it
$phone_ip = shift;
$local_ip = shift;
$sip_to = shift;
$sip_from = "0";
$tm = time();
$call_id = $tm . "msgto$sip_to";
$httptime = `date -R`;
$MESG = "NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP $local_ip
From:
To:
Event: check-sync
Date: $httptime
Call-ID: [EMAIL PROTECTED]
CSeq: 1300 NOTIFY
Contact:
Content-Length: 0
";
$proto = getprotobyname('udp');
socket( SOCKET, PF_INET, SOCK_DGRAM, $proto );
$iaddr = inet_aton("$phone_ip");
$paddr = sockaddr_in( 5060, $iaddr );
bind( SOCKET, $paddr );
$port = 5060;
$hisiaddr = inet_aton($phone_ip);
$hispaddr = sockaddr_in( $port, $hisiaddr );
if ( send( SOCKET, $MESG, 0, $hispaddr ) ) {
print "reboot of phone ", "$phone_ip", " was successful", "\n";
}
else { print "reboot of phone ", "$phone_ip", " failed", "\n"; }
}
exit;
Enjoy,
John
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[Asterisk-Users] TE410P/Zaptel
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, How can I configure the TE410P card to act as master instead of slave? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAESr02TEAILET3McRApNJAJ9vckuAym/j+d/19qmoNtE9adOESACeNfti CXt25JHMMqvIlZGvkqxJwSs= =BBRR -END PGP SIGNATURE- ___ 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] Digium X100P for $43
> When you see 'X100P' cards for that price you can be assured that they > are either used Digium (who would want to part with their X100P??) or > more likely (as in this case and most cases) manufactured by a > third-party. I don't know if they are technically 'pirate' cards, but I > wouldn't recommend purchasing one. Stick to the Digium hardware that you > know will work with Asterisk. As an added benefit you will be helping to > support a company that supports Asterisk. I have an X101P I'm getting rid of (no PSTN in this house!) -- but just to clarify, digium does not manufacture their own X101P cards. The others (T100P/TDM400P/TE410P/etc.) are actual genuine digium/OSS products. It appears that the X101P is just a specific winmodem that they've standardized on. I have no issues paying digium since they give full support for these cards. Buy it outside and you've got no support. If that's not an issue for you, buy the cheapies. I encourage you as much as the next guy to support Digium but I'm not going to villianize anyone who decides to save some bucks. That's exactly what Digium's done, too. :-) Regards, Andrew ___ 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
