[Asterisk-Users] Hop-On WIFI Phone MSRP $40
I have a lot of folks asking me about an auto-negotiating WLAN phone supposedly being brought to market by Hop-On, which is touted to carry an MSRP of $40 Press photos (stock art) of the device shows it looks almost identical to devices from Zyxel and UTStarCom. I am trying to explain to folks there is no way in hell you are going to be able to buy these phones, hardware only, for $40 Hop-On's press releases are somewhat ambiguous, but at this price point, they would have to bundle the phone with a length service contract in order to subsidize the hardware cost on the phone. Anyone have any inside info regarding Hop-On? Cory Andrews Partner / Purchasing VOIPSupply.com ++ 454 Sonwil Drive Buffalo, NY 14225 ++ v - 800.398.VOIP Ext 22 f - 716.630.1548 e - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth?
You can set, in the linksys, the codec G729 for your line. In the Linksys also set only to use that codec. This can be done at the admin page of the line you use in the linksys. Also do that in the asterisk for your device. First buy the license from Digium. Then you will use less bandwidth and have a better sound upstream. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 1:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth? Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to the same hotel, I can get reliable connectivity. Assuming the hotel isn't helping me on the QOS front, and the Hotel's connectivity is the last word, then my Vonage ATA should be choppy, as well, no? This is what leads me to think I can do some tweaking later, Paul - Original Message - From: Greg Oliver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:17 PM Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth? Nothing you can do on this one.. Without the provider accepting your QoS settings, you are at their mercy. And yes, you are correct, most multi-tenant dwellings use xDSL for their connectivity due to it's price, and the upstream is usually less bandwidth than the downstream.. -Greg On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote: So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP session between two end users
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Tuesday, June 28, 2005 6:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users Erdem HAKİ wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Monday, June 27, 2005 8:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users Erdem HAKİ wrote: Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used canreinvite=yes but it didn't work. Description from asterisk conf. File; (canreinvite=yes; allow RTP voice traffic to bypass Asterisk) It's sip.conf. reinvites only work if the codec is the same for the two endpoints and Asterisk does NOT have to listen for DTMF (no t or T on the dial line, no meetme, etc.) *** We use same codec and don't use meetme etc... So what else should i do? How are you determining if RTP audio is going thru Asterisk? Remember, SIP signaling will always go thru Asterisk. Also do a sip show channels during a call to confirm that the codecs are the same. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I determine signaling with ethereal and i am sure that both sides use the same codec. By the way, i searched forum again and i read something below; In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtps are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. As bkw pointed out, use canreinvite=yes for each sip phone definition. But, that will only work if the phones can reach each other directly (the phones and/or asterisk can't be behind a nat/firewall box). Thanks Erdem HAKI [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing Calls
HI! I configured asterisk to send all outgoing calls to our Gateway. I noticed when asterisk sends call to gateway that he represents all calls as asterisk and not as callerID(number of sjphone client registerd to asterisk). Can anyone give me an example of such configuration? Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI
Howdy, Am Dienstag, den 28.06.2005, 09:01 +0200 schrieb vdasilva: Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN card with the CAPI drivers... Yes, BRIstuff and the hfc-pci will provide echo cancelation. With the Fritz card however you will NOT get echo cacnelation. I have tried using bristuff on RH9 but couldn't get my zaptel to compile... Do you have _configured_ kernel sources installed? If you run a 2.6 kernel do you have the necessary scripts to build kernel modules (these are built during the kernel compilation process)? Then there is the issue of timing, ztdummy or zaprtcand QoS setup on the Linux box... Can anyone who has a 100% working Asterisk implementation using any of the techniques described above tell me more... I will happily upgrade to the Fritz card if it will solve all the problems... Thanks Vicente best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using SipP to produce RTP load?
That would probably be me. You could use a lot of different things to do the testing, one would be the tcl script in your asterisk/contrib/scripts directory, some more can be found in the beginning of this presentation: http://astertest.com/astricon_performance.ppt We started some callgenerator for asterisk a very long time ago. ( I have to admit its far from ready and contains many bugs). A howto for this tool can be found at : http://www.asteriskguru.com/tutorials/astertest.html If you want to use sipp, be sure to use playback on your asterisk server and not app_milliwatt, meetme or echo. (those applications will not send any rtp if nothing was received). SIPP only does echoing Zoa --- Asteriskguru tutorials: http://www.asteriskguru.com/tutorials/ Visit ClueCon - the asterisk developpers conference: http://www.cluecon.com/ - Dates: August 3, 4 and 5, 2005 Best Western Chicago West Matthew Boehm wrote: Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729 conversion would occur. Nothing. I was able to pump 10 simul calls that went this path: sipp - asterisk - pri - telco -pri -asterisk ..and still no 729 usage or any other discernable load on the server. Can anyone offer suggestion on how to really simulate calls (using sipp or other tester) to asterisk to verify its ability to process X calls? I know someone out there has done this, but forget who it was. Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simultaneus calls?
Thanks for your help Bernard, it's realy useful web site, but i also want to know limits which depens on hardware of the box. Any practical experience? Thanks again :-) Erdem HAKI - [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernard Cresencia Sent: Tuesday, June 28, 2005 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] simultaneus calls? I did a google search on 'voip speed test' - the first site is very good. Here's the link: http://www.talkswitch.com/voip/voip_test.php It will test both your download and upload speeds and will let you know how many concurrent calls at different codecs your connection will support. Try it a few times and on different times of the day to get an average. --- Erdem HAKÝ [EMAIL PROTECTED] wrote: Yes it is DSL and outbound speed is aslo 1Mbit, it's a dedicated server and we just use to talk. I look at the web site which you suggested, but i want to learn how many calls supported practically? Any information do you have? Thanks Erdem HAKI - [EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 28, 2005 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] simultaneus calls? The 1 m internet connection will be the limiting factor in your setup, you did not state what type of internet connection, but given the speed of 1 mbit it must be DSL (or maybe fraction t/e1). Is the outbound speed also 1m? Is there data on the line also? How much? What about voice Qos? You should start here http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKI Sent: Tuesday, June 28, 2005 3:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] simultaneus calls? Hello, How can i learn my asterisk how many simulyaneus calls support? My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit internet connection. Thanks for your interest... Erdem HAKI - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns 4 port BRI problem
Hi, CRC errors are caused by bit errors on layer 1. In most cases this is a cable issue. Did you try replacing the cable from the NT1 to the quadBRI? How long is that cable? However if only 1 of the 2 B channels are working then you might have your BRI lines get checked or try a different ISDN device on those lines. best regards Klaus -- Klaus-Peter Junghanns Am Dienstag, den 28.06.2005, 18:22 +0200 schrieb Doug Reid - Stormcorp: Hi All I have a Junghanns BRI 4 port installed where only the first channel of each line is working i.e. channels 1 and 4 work but 2 and 5 don't. Our config is the same on this box as 15 other similar installations where all works well. the only error I see is in /var/log/messages: Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:51:27 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:53:09 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:56:48 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:58:06 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 16:01:01 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Can anyone help with this? Thanks Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Correction to Janghanns BRI problem
Hi, what signalling does the telco run on those lines? best regards Klaus Am Dienstag, den 28.06.2005, 19:02 +0200 schrieb Doug Reid - Stormcorp: Hi all Correction on my last mail, I found that line 1 both channels work but on line 2 none work. I have 2 BRI ISDN lines coming in on port 1 and 2 (4 channels) on a Junghanns 4 port. The setup by the Telco on this ISDN is different than our others, they have 2 lines (4 channels) that are all connected to one telephone number i.e. 701 5161. The second number should be 701 5162 but this number does not exist. If we put a Sirrix card in all 4 channels (2 x BRI) work fine on 701 5161 but when we put a Junghanns in only one line works. It seems like the second line is not given a B channel from the NTU side of the Telco. Error in /var/log/messages: Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:51:27 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:53:09 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:56:48 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Please if anyone could suggest a fix here it would be much appreciated. Thanks Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using SipP to produce RTP load?
On 29 Jun 2005, at 04:51, Matthew Boehm wrote: Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729 conversion would occur. Nothing. I was able to pump 10 simul calls that went this path: sipp - asterisk - pri - telco -pri -asterisk ..and still no 729 usage or any other discernable load on the server. Can anyone offer suggestion on how to really simulate calls (using sipp or other tester) to asterisk to verify its ability to process X calls? I know someone out there has done this, but forget who it was. I think you mean Signate. I saw a presentation at Astrcon . They call the milliwatt generator to fill the RTP stream. They were getting 122 passthrough ulaw calls on a 'stock' pc. If I remember right the benchmark scripts and methodology are available. If you are looking to benchmark that 4 way 500Mhz box of yours I'd be _very_ interested in the results with varying numbers of CPUs. Signate were saying that the limiting factor (with ulaw passthrough) is the PC architecture (bus and interrupt structure) not the CPU. I've done a _tiny_ experiment myself. I found that a single 729-alaw- PRI call uses less than 10% of the CPU on a 1Ghz nemiah Tim. Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Good soft-phone on Linux
* Hamish Whittal ([EMAIL PROTECTED]) ha scritto: Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. it may be more of what you need but using asterisk with the OSS/Alsa module turns it in a very efficient client (it can run also without X installed ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using SipP to produce RTP load?
On 29 Jun 2005, at 04:51, Matthew Boehm wrote: Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729 conversion would occur. Nothing. I was able to pump 10 simul calls that went this path: sipp - asterisk - pri - telco -pri -asterisk ..and still no 729 usage or any other discernable load on the server. Can anyone offer suggestion on how to really simulate calls (using sipp or other tester) to asterisk to verify its ability to process X calls? I know someone out there has done this, but forget who it was. I think you mean Signate. I saw a presentation at Astrcon . They call the milliwatt generator to fill the RTP stream. They were getting 122 passthrough ulaw calls on a 'stock' pc. If I remember right the benchmark scripts and methodology are available. If you are looking to benchmark that 4 way 500Mhz box of yours I'd be _very_ interested in the results with varying numbers of CPUs. Signate were saying that the limiting factor (with ulaw passthrough) is the PC architecture (bus and interrupt structure) not the CPU. I've done a _tiny_ experiment myself. I found that a single 729-alaw- PRI call uses less than 10% of the CPU on a 1Ghz nemiah Tim. Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID Bug?
hi, all: I have two phones, one is SIP/200, another is IAX2/203. Now, i use IAX2/203 call to SIP/200, sometime CallerID display is 203(at phone SIP/200), sometime display is 200. Is this a bug? Please help me! Sorry my english. Li Yuqian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ERROR[22927]: Failed to create socketpairfor player(24, Too many open files).
Thanks for the reply. But how do you troubleshoot which application is the culprit. Any ideas ? I am using FEDORA 3. Rgds T.E.Yap ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC not billing
That looks right, the database is being updated properly. The last call
lasted 9 seconds and cost you 1.5c so it should show up in the database.
You did create a 2-digit card called '21' right?
- Original Message -
From: Juan Luis Moyano [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 29, 2005 12:13 AM
Subject: Re: [Asterisk-Users] ASTCC not billing
Bernard Cresencia wrote:
sorry, I meant my.cnf, not my.conf.
Once logging is enabled, I would do tail -f
/var/log/myslqd.log and watch as the database is being
accessed during a call.
I've done what Bernard suggested and this is my output from mysql.log on
a successful call to number 612 on FWD. I'd like to know if any of you
see something wrong or rare. Thanks a lot.
Time Id CommandArgument
050629 1:02:02 1 Connect [EMAIL PROTECTED] on astcc
050629 1:02:04 1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query UPDATE cards SET used='1801' WHERE
number='21'
1 Query UPDATE cards SET inuse='1' WHERE
number='21'
050629 1:02:10 1 Query SELECT * FROM routes WHERE '612'
RLIKE pattern ORDER BY LENGTH(pattern) DESC
050629 1:02:25 1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM trunks WHERE name='FWD'
050629 1:02:37 1 Query INSERT INTO cdrs
(cardnum,callerid,callednum,trunk,disposition,billseconds,billcost,callstart)
VALUES ('21', '\Coco\ 21', '612', 'FWD', 'ANSWER', '9', '150', 'Wed
Jun 29 01:02:37 2005')
1 Query UPDATE cards SET used='1951' WHERE
number='21'
1 Query UPDATE cards SET inuse='0' WHERE
number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query UPDATE cards SET used='1951' WHERE
number='21'
1 Query UPDATE cards SET inuse='0' WHERE
number='21'
1 Quit
--
Juan Luis Moyano
[EMAIL PROTECTED]
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[Asterisk-Users] chan_capi-cm-0.5.3 fixup release
Hi all, on sourceforge.net I added the fixup release 0.5.3 of chan_capi-cm driver. The changes from 0.5.2 to 0.5.3 are: - voice data queue (send buffer) fix - fix for CVS-HEAD of Asterisk (Thanks to Frank Sautter) I have tested this version with Asterisk 1.0.7, 1.0.8 and HEAD(2005/06/28). Have fun Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM Hunting
Hi, Need to implement hunting (create a hunt group so my subscribers can have a single GSM number for access to me)of GSM SIMs on a GSM bank independent of the Telco for the SIMs. Anyone got an EXACT idea how to do this? Thanks, Latex. Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with chan_capi 0.3.5 , divactrl, eicon diva server, and kernel 2.6.10/2.6.12
On Tue, 28 Jun 2005, Luis Vazquez wrote: Hello all, I'm having problems getting chan_capi 0.3.5 to work well with an Eicon Diva Server card using using the driver from linux kernel both 2.6.10 and 2.6.12 (vanilla versions). Have you tried the chan_capi-cm version from sourceforge ? I have a (really two) producción system(s) running chan_capi in another identical Eicon Card using kernel 2.4.27 and the Diva Server drivers from Eicon. I installed an compiled the source level rpm divas4linux_EICON-104.429-1.i386.rpm into a binary rpm named divas4linux_EICON-105.465-1.i386.rpm. It works (allmost) without problems except for the fact of some random segmentation faults in Asterisk when capi is handling incoming calls (the same problem has already been reported in the list without solutions as long as I know). This doesn't sound like a problem with the diva drivers... The problem with Eicon's Diva Server Driver is they dont give a version to work with kernel 2.6 and I need to upgrade to k2.6 for many reasons, besides I prefer to use the open source version. You don't want to use the drivers which are part of kernel 2.6.x ? 1- When I made a call from Asterisk through the capi interface it take many seconds (around 10 secs) to get connected and then you hear a static noise, something like permanent random clicks mixed with the voice. I used the ditrace tool to debug a call and I see many Layer 1 - [Lost Framing] in the output, not present when using the same BRI line in the production PBX using k2.4.27 and Eicon's Driver/Tools. I have tried exchanging the cards too, but the problem is the same. ... 0:01:53.032 s 2 Layer 1 - [Lost Framing] - 0:01:53.032 s 2 Layer 2 - [Idle] 0:01:53.034 s 2 Layer 1 - [Syncronized] 0:01:53.034 s 2 Layer 2 - [Idle] This looks like a layer 1 connection problem. What type of ISDN line do you have and which divactrl load parameters did you use? 2 - After the call is terminated from asterisk side I see the hangup on asterisk CLI but the capi line keeps busy for a while (at least 15-20 secs). I see the orange light in Eicon Card is on and I'm unable to place a new call in meantime. This might be the same like 1. 3 - Sometimes when I take the system down (halt or reboot) I get a kernel panic with null pointer errors related to capi drivers. Backtrace / Oops message ? What is the best isdn libs version for using with chan_capi, k2.6.* and divactrl?? your versions should be okay. Does anybody has a working chan_capi environment with a recent (2.6.10 up) kernel, and could give a hint on any of those synthoms?? I would like to try the new sourceforge chan_capi version 0.5.*, but it is not clear if these works with asterisk stable or only with cvs version? If so, are they (mostly) stable enough for testing in a production system (compared to chan_capi 0.3.5)? It does work with stable and HEAD of Asterisk. The latest changes to the voice buffer caused some problems, but 0.5.3 should be working (at least here in my environment). Armin___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC not billing
The reason for the problem is clear below, theASTERISKCDRDB database is being updatedinstead of theASTCCDB database which holds the cdrs and BILLCOST.
How can this problem be corrected???
3 Query UPDATE cards SET used='0' WHERE number='58767059' 3 Query UPDATE cards SET inuse='0' WHERE number='58767059' 3 Query SELECT * FROM cards WHERE number='58767059' 3 Query UPDATE cards SET used='0' WHERE number='58767059' 3 Query UPDATE cards SET inuse='0' WHERE number='58767059' 3 Quit 4 Connect [EMAIL PROTECTED] on asteriskcdrdb 4 Query INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-06-29 05:17:42','\"Mike\" 1234','1234','777','TEST', 'SIP/1234-f285','SIP/213.45.62.117-1732','Dial','SIP/213.45.62.117/19315461298|30|HL(6:6:3)',42,42,'ANSWERED',3,'')Juan Luis Mo
yano
[EMAIL PROTECTED] wrote:
Bernard Cresencia wrote: sorry, I meant my.cnf, not my.conf. Once logging is enabled, I would do tail -f /var/log/myslqd.log and watch as the database is being accessed during a call.I've done what Bernard suggested and this is my output from mysql.log ona successful call to number 612 on FWD. I'd like to know if any of yousee something wrong or rare. Thanks a lot.Time Id Command Argument050629 1:02:02 1 Connect [EMAIL PROTECTED] on astcc050629 1:02:04 1 Query SELECT * FROM cards WHERE number='21'1 Query SELECT * FROM cards WHERE number='21'1 Query SELECT * FROM cards WHERE number='21'1 Query SELECT * FROM cards WHERE number='21'1 Query UPDATE cards SET used='1801' WHEREnumber='21'1 Query UPDATE cards SET inuse='1' WHEREnumber='21'050629 1:02:10 1 Query SELECT * FROM routes
WHERE
'612'RLIKE pattern ORDER BY LENGTH(pattern) DESC050629 1:02:25 1 Query SELECT * FROM cards WHERE number='21'1 Query SELECT * FROM trunks WHERE name='FWD'050629 1:02:37 1 Query INSERT INTO cdrs(cardnum,callerid,callednum,trunk,disposition,billseconds,billcost,callstart)VALUES ('21', '\"Coco\" 21', '612', 'FWD', 'ANSWER', '9', '150', 'WedJun 29 01:02:37 2005')1 Query UPDATE cards SET used='1951' WHEREnumber='21'1 Query UPDATE cards SET inuse='0' WHEREnumber='21'1 Query SELECT * FROM cards WHERE number='21'1 Query UPDATE cards SET used='1951' WHEREnumber='21'1 Query UPDATE cards SET inuse='0' WHEREnumber='21'1 Quit-- Juan Luis Moyano[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
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RE: [Asterisk-Users] GSM Hunting
Hi, -Original Message- Need to implement hunting (create a hunt group so my subscribers can have a single GSM number for access to me)of GSM SIMs on a GSM bank independent of the Telco for the SIMs. Anyone got an EXACT idea how to do this? If you want 1 GSM number that can access many GSM SIM's you need assistance of the GSM telco. Alternatively you could enable call forward on busy for all the SIM's, so they daisy chain through there. Might end up being a very expensive solution though... Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Teliax Problems
I'm currently unable to register with Teliax's server via IAX2 and can't reach them via either of their phone numbers. Their website is up and I have logged a support incident. Is anyone else experiencing the same problems? Having been caught up in the Broadvoice fiasco a couple of months back, I'm hoping that Teliax is not going through the same sort of thing. Malcolm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI and Caller ID name not showing.
I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4 port Eicon Diva card. All works fine, but i'd like calls from the PBX to Asterisk to show the Caller ID name and not just the number. I know this information is being presented by looking through the ISDN trace for the Eicon Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is '605'. Can anyone point me in the right direction to get this sorted?. It's works with X100P cards :) _ Winks nudges are here - download MSN Messenger 7.0 today! http://messenger.msn.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_conference in dial plan?
Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferencing... ; Make as many of these contexts as you have seperate conference bridges ; change conferencename in each [conf-conferencename] exten = join,1,System(/opt/asterisk/bin/conference-announce conferencename in) exten = join,2,Conference(conferencename/S/1) exten = h,1,System(/opt/asterisk/bin/conference-announce conferencename out) [confhelper] ; make one of these extensions per seperate conference bridge exten = conf-conferencename,1,Conference(conferencename/S/1) exten = in,1,Answer() ; if I use Playback here instead of BackGround, asterisk crashes exten = in,2,BackGround(conf-announce) exten = in,3,ResponseTimeout(5) exten = in,4,Hangup() exten = out,1,Answer() exten = out,2,BackGround(conf-leave) exten = out,3,ResponseTimeout(5) exten = out,4,Hangup() how do I setup up app_conference to respond to an extention? Just a real simple example to get me started would be appreciated... I've tried a few things along the lines of the example meetme extention ie exten = 901,1,app_conference(901||1234) or exten = 901,1,cmd_conference(901||1234) But I guess its expecting too much to think that this would fireup app_conference Thanks in advance for any help. Cheers, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Problems
I'm currently unable to register with Teliax's server via IAX2 and can't reach them via either of their phone numbers. Their website is up and I have logged a support incident. Is anyone else experiencing the same problems? Having been caught up in the Broadvoice fiasco a couple of months back, I'm hoping that Teliax is not going through the same sort of thing. An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_rtp_read: Unknown RTP codec 100 received21 when receiving fax
I'm testing NVBackgroundDetect with Sipura-300 and I get this error: rtp.c:505 ast_rtp_read: Unknown RTP codec 100 received21 Does anybody know what is it? -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommend against Teliax as primary ITSP
I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip- PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of service they do. I have been using Teliax for about 3 months now, and they were my first ITSP when I started playing with Asterisk and my Grandstream BT101. I picked them arbitrarily because they had low rates, and supported the IAX2 protocol, which I determined to be more firewall friendly. Right away, I was happy with the reponse, the online ordering, and the low rate. It didn't take long for the multitude of outages to occur. Now, while I claim to be no VoIP expert, I did a variety of tests to make sure the problems weren't on my end. I recommended Teliax to a business partner, who has a Linux box in a data center downtown, and had access to their system as well. When I'd find outages, I would first check to see if they were having the same problem. So far, every time I had a problem, so did they. I am also registered with FWD on IAX2 and they were always up. Any tech support calls to Teliax would take more than 2 days to get a response. Only when I threatened to leave would someone suddenly pop up and answer my concerns. They claim to have been changing bandwidth providers (away from rockynet, or at least companies that peer with Cogent), but traceroutes show they are still with them. So far, when I've actually gotten ahold of a tech support person, they have told me to try different addresses for the server. They've changed recently from voip.teliax.com to ast01.teliax.com to voip-co1.teliax.com. Guess what? All the same server. Its just more of the same runaround. Since the day I switched to VoIP (with Teliax) as my primary outbound calling, more people have laughed at me for my choice of VoIP as a telco medium than can be counted. And these are people who respect me in the Telco community, and who I have been trying to convince of the benefits. They don't see the benefits when they can't call me, and I can't call them. I understand that all companies have their problems, especially with such emerging technology as VoIP. I would have very little problem with Teliax, and use a secondary provider as a backup, if they were more forthright in explaining their problems, and notify their customer base within a reasonable time when they are going to have outages due to network changes. As it stands, I now have to find a new provider that will at least duplicate the features of Teliax. The hardest part of this is, they offer BYOD, use IAX2, and let you change your callerid presentation. These are all things that I MUST have. If anyone has some positive results with a similar competitor, I'd love to hear about it. In the meantime, I have to change back over to PSTN lines temporarily, since I can't rely on service from Teliax. I hope any/all of you that use their service have better luck than I have with them. Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Problems
Lets not jump the gun here..one failed iax registration does not a bankrupt company make... (p.s., yes my registrations are not getting responses either) Mark On 6/29/05, Rich Adamson [EMAIL PROTECTED] wrote: I'm currently unable to register with Teliax's server via IAX2 and can't reach them via either of their phone numbers. Their website is up and I have logged a support incident. Is anyone else experiencing the same problems? Having been caught up in the Broadvoice fiasco a couple of months back, I'm hoping that Teliax is not going through the same sort of thing. An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax Problems
Try voip-co2.teliax.com to register with. And read my other letter I suppose. This domain is apparently working as of 4:30, but have had the same problem since 1:30 AM PDT. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Malcolm Taylor |Sent: Wednesday, June 29, 2005 4:14 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Teliax Problems | | | |I'm currently unable to register with Teliax's server via IAX2 |and can't reach them via either of their phone numbers. Their |website is up and I have logged a support incident. | |Is anyone else experiencing the same problems? Having been |caught up in the Broadvoice fiasco a couple of months back, |I'm hoping that Teliax is not going through the same sort of thing. | |Malcolm | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax Problems
Does anyone have anything +/- to say about TeleSIP? They appear to have local DIDs where I live and all comments on the wiki indicate they are reputable.. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Wednesday, June 29, 2005 5:22 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Teliax Problems | | | I'm currently unable to register with Teliax's server via IAX2 and | can't reach them via either of their phone numbers. Their |website is | up and I have logged a support incident. | | Is anyone else experiencing the same problems? Having been |caught up | in the Broadvoice fiasco a couple of months back, I'm hoping that | Teliax is not going through the same sort of thing. | |An ethereal trace indicates the IP address is active, but it |is not responding to iax packets (registration). So, either |their asterisk app has failed or they have folded their tent as well. | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Equipment for small office setup
Hi there... I've to setup an Asterisk system for a small office, I haven't done one of these in at least a year and was wondering if someone could just let me know what sort of phones are doing well these days. It just needs 9 phones in the office, for general use, no fancy things required for that, just accept calls, transfer calls etc. 1 Master phone for a receptionist. Is there an easy way at the moment for one of these bigger phones (cisco or whatever) to view the status of the various lines etc? Some phone with an expansion board maybe? Finally, I'm still debating whether to use 2 analogue lines or 1 ISDN. 2 analogues would be cheaper I believe, what sort of ISDN card works well with Asterisk? Thanks for any help! -- Steve Foy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC not billing
If astccdb exists, go to the database configuration
page [Configure] and change the database name to the
correct one. You may have to set up permissions on
this database if it wasn't set up before. If it
doesn't exist, use the 'Create Database' button to
create a new one.
--- Ade Agbero [EMAIL PROTECTED] wrote:
The reason for the problem is clear below, the
ASTERISKCDRDB database is being updated instead of
the ASTCCDB database which holds the cdrs and
BILLCOST.
How can this problem be corrected???
3 Query UPDATE cards SET used='0' WHERE
number='58767059'
3 Query UPDATE cards SET
inuse='0' WHERE number='58767059'
3 Query SELECT * FROM cards
WHERE number='58767059'
3 Query UPDATE cards SET
used='0' WHERE number='58767059'
3 Query UPDATE cards SET
inuse='0' WHERE number='58767059'
3 Quit
4 Connect
[EMAIL PROTECTED] on asteriskcdrdb
4 Query INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2005-06-29 05:17:42','\Mike\
1234','1234','777','TEST',
'SIP/1234-f285','SIP/213.45.62.117-1732','Dial','SIP/213.45.62.117/19315461298|30|HL(6:6:3)',42,42,'ANSWERED',3,'')
Juan Luis Moyano [EMAIL PROTECTED]
wrote:
Bernard Cresencia wrote:
sorry, I meant my.cnf, not my.conf.
Once logging is enabled, I would do tail -f
/var/log/myslqd.log and watch as the database is
being
accessed during a call.
I've done what Bernard suggested and this is my
output from mysql.log on
a successful call to number 612 on FWD. I'd like to
know if any of you
see something wrong or rare. Thanks a lot.
Time Id Command Argument
050629 1:02:02 1 Connect [EMAIL PROTECTED] on astcc
050629 1:02:04 1 Query SELECT * FROM cards WHERE
number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query UPDATE cards SET used='1801' WHERE
number='21'
1 Query UPDATE cards SET inuse='1' WHERE
number='21'
050629 1:02:10 1 Query SELECT * FROM routes WHERE
'612'
RLIKE pattern ORDER BY LENGTH(pattern) DESC
050629 1:02:25 1 Query SELECT * FROM cards WHERE
number='21'
1 Query SELECT * FROM trunks WHERE name='FWD'
050629 1:02:37 1 Query INSERT INTO cdrs
(cardnum,callerid,callednum,trunk,disposition,billseconds,billcost,callstart)
VALUES ('21', '\Coco\ 21', '612', 'FWD',
'ANSWER', '9', '150', 'Wed
Jun 29 01:02:37 2005')
1 Query UPDATE cards SET used='1951' WHERE
number='21'
1 Query UPDATE cards SET inuse='0' WHERE
number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query UPDATE cards SET used='1951' WHERE
number='21'
1 Query UPDATE cards SET inuse='0' WHERE
number='21'
1 Quit
--
Juan Luis Moyano
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Teliax Problems
An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. I am sure it's just a crashed server, wait an hour and let the support people deal with it. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing
You would be better using extensions_custom only because of the fact that when you restart ampportal, it will overwrite extensions_additional with what ever it has stored in the Database. I've actually taken to adding the code that I build onto what AMP generates into the database. For example I had to add in a default option for a Digital Receptionist I used the phpMyAdmin that's installed with [EMAIL PROTECTED] and inserted the data into the extensions table. Hatton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Problems
On Wed, 29 Jun 2005 08:15:20 -0400 Chris Mason (Lists) [EMAIL PROTECTED] wrote: An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. I am sure it's just a crashed server, wait an hour and let the support people deal with it. -- Chris Mason NetConcepts The server is up as IAXPing generates responses from voip-teliax.com and voip-co2.teliax.com Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Shoutcast Music On Hold problems?
bash-3.00# cat musiconhold.conf | more ; ; Music on hold class definitions ; [classes] ; Christian Rock.NET ;default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ;loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ; Cleft in the Rock Radio (TESTING) default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ bash-3.00# pwd /var/lib/asterisk/mohmp3-empty bash-3.00# ls -la total 8 drwxr-xr-x 2 root root 4096 Jun 15 15:21 . drwxr-xr-x 9 root root 4096 Jun 15 15:18 .. -rw-r--r-- 1 root root0 Jun 15 15:21 empty.mp3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, June 29, 2005 1:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Fw: [Asterisk-Users] Shoutcast Music On Hold problems? - Original Message - From: hank [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I am using [EMAIL PROTECTED] 1.0 my mp3 is called mp3 it has nothing before it it is 0 bytes does my mp3 of 0 bytes need to have a .mp3 or does it need to be called anything? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 11:52 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? Worked for me with a different stream... I ran into this same problem before - but it was my own fault for not RTM... Both the manual and ast install advised of verifying correct version of mpg123... I had wrong version and thus got no noise... If you follow the directions explicitly laid out on the wiki you should have no problems. I use christianrock.net's shoutcast stream Like this in musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Tuesday, June 28, 2005 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1
Title: Message I receive this error on the asterisk console and it is pretty much ALWAYS coming up. Sometimes there will be a break where it does not display. We had our PRI provider test the lines and they claim that there is no signalling problem. It doesn't matter if there are no calls or if there are 10 calls in progress the error is still displayed. I also get an annoying popping or clicking sound but that doesn't always correspond with this error coming up so it is likely a separate issue. I have loaded all modulesby hand like below as someone suggested in a search for HDLCerrorson the list. insmod zaptel insmod wct1xxp Unfortunately it did not help Has anyone run into this in the past? Michael ;zapata.conf switchtype=nationalcontext=incoming_eli_pri_1signalling=pri_cpegroup=1channel = 1-11bchan=1-11dchan=24 ;zaptel.conf span=1,1,0,esf,b8zsbchan=1-11dchan=24 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got
[Asterisk-Users] Machine Sizing
Hi, I am planning to try Asterisk and would like some guidelines on the size of machine I need. Is there a page somewhere with some suggestions? Kevin Roche ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration
On Sat, Jun 25, 2005 at 07:58:24PM -0500, Greg Oliver wrote: That works well. You may also want to make sure your compatibility matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more issues than I care to talk about. The GNUGk web site has the best matrix to follow.. Do you have a specific URL, the only thing I can find is http://www.gnugk.org/interoperability.html, which doesn't sound exactly like what you're talking about. Thanks, Barney. -- Barney Sowood [EMAIL PROTECTED] Tel: +44 (0)845 226 5841 Sowood Co Ltd, 22 Manor Place, Edinburgh, EH3 7DS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Connecting PBX to Asterisk
Title: [Asterisk-Users] Problem with Connecting PBX to Asterisk The framing is ESF/B8ZS. But I have had some luck and have gotten to the point where when dialed from Asterisk, the digits reach the telrad switch and the DID that I have configured in the telrad switch works and rings the right extensions. However, when I do the reverse, there are NO digits reaching the asterisk. Finally things worked by using em in zaptel.conf and em_w in Zapata.conf. I would appreciate if someone can help me figure out what could be the problem in receiving the digits from the telrad switch/pbx. I am very sure that the telrad part is working fine as I see that when I dial out, it picks up a trunk channel from the T1 connected to asterisk and dials. So it is to do with the asterisk part not reading the digits. Thanks -- What about framing? ESF/B8ZS vs D4/AMI? -Original Message- From: Karthik Natarajan Sent: Wed, June 22, 2005 10:01 am As suggested by you, I have tried the timing parameter with 0 an 1 both but to no avail. Signalling I have tried are both em_w and featd (suggested by digium tech support). Zap show channels shows all 24 channels to be ok with no alarms. Zttool also confirms that. The problem is that the pbx (telrad) does not even seem to sense the T1 that is plugged in. No yellow and no green. Only the RED LED glows. But when I try a T1 loopback plugged into that the card turns green on telrad suggesting that the card is fine. (T1 loopback build using a small connector with 1 to 4 and 2 to 5 connected). Also regarding timing, I do have 2 X100Ps installed so I wonder if this needs to be primary source of timing? As I mentioned earlier I am getting a GREEN on asterisk side (TE405P card) while a red on the pbx side (TELRAD) Any thoughts? Clock source will be important here. For phase one, you should probably set asterisk to time from the PBX since the PBX is likely timing from the T1 circuit. At phase two, you will likely want to reverse this having your PBX clock from the Asterisk system and having Asterisk clock from the telco T1. This of course assumes that there is only 1 Telco T1 involved. Timing is the first most important consideration. After that, you want to verify that both ends are using the same signalling type (it appears that you are using CAS signalling). Check that your PBX is using CAS and find out exactly what type. Zapata will need to be configured to use the same type of signalling. Check both sides looking for any red alarms etc that might indicate a cable problem. From the CLI, 'zap show channels' or using zttool from the command line should help you determine the status of the link. Red alarms usually indicate that this end sees an out of frame condition while a yellow means that the opposite side sees and out of frame. A yellow on one side is a red on the other. Karthik Natarajan InfoPro Corporation 732-283-2589 x 241 karthik at infoprocorp.com ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Machine Sizing
Kevin Roche wrote: Hi, I am planning to try Asterisk and would like some guidelines on the size of machine I need. Is there a page somewhere with some suggestions? http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Revision I Board TDM04b
On Tuesday 28 June 2005 23:15, Rich Adamson wrote: I cannot get this thing to work. Anyone know of any tricks? Call digium support; its free. Well technically it's not free. You just paid for support in the price of the card (of all their cards)... -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax Problems
It's up and running again now. I just found it a little disconcerting not to be unable to reach their support numbers during the outage. Malcolm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, June 29, 2005 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Teliax Problems An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. I am sure it's just a crashed server, wait an hour and let the support people deal with it. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Caller ID name not showing.
On Wed, 29 Jun 2005, louis g wrote: I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4 port Eicon Diva card. All works fine, but i'd like calls from the PBX to Asterisk to show the Caller ID name and not just the number. I know this information is being presented by looking through the ISDN trace for the Eicon Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is '605'. Can anyone point me in the right direction to get this sorted?. It's works with X100P cards :) What 'name' do you mean? Is it a subaddress? Please paste an example for that Eicon card trace where you see that name. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Looks like 9 out of 10 calls are failing on voipjet at the moment (at least terminating to South Florida numbers). Keep getting message that says number can not be completed as dialed. Anyone else seeing this? On 6/15/05, Pedro [EMAIL PROTECTED] wrote: Couple of days. Apparently the new US carrier has some changes that needs to be made. On 6/14/05, Wiley Siler [EMAIL PROTECTED] wrote: Did they say when it would be corrected? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, June 14, 2005 9:22 AM To: Matt Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? Caller ID is still not working to certain areas. This problem was confirmed by voipjet tech support in their last e-mail to me. On 6/13/05, Matt [EMAIL PROTECTED] wrote: I never noticed any problems.. so I can't comment :) hehe On 6/11/05, Pedro [EMAIL PROTECTED] wrote: Finally got a response from voipjet support and they say they have switched to a new provider for US termination. I have yet to test this out as I have not had a chance to build them back into our routes but will report my findings once I do. Anyone else notice any improvements? On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hop-On WIFI Phone MSRP $40
Unfortunately no. Someone say the press release and bugged me about it as well but I haven't seen anything that would indicate they plan on doing anything more than parting with carriers with large rollouts of these phones. That MSRP seems too good to be reality too. -- William ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM card and voicemail volume
-Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] I was able to raise the volume from inaudible to acceptable by increasing the RxGain in zapata.conf by 5db. I'd rather not go the uncomressed wav route, as it will chew up storage in my email system. This is an acceptable work-around if you're just doing voicemail and IVR. It may cause echo or excessive volume levels if you're also doing regular calls, though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App_conference in dial plan?
exten = 901,1,Conference(Internal Test Conference/S/1) Looks like it does the job... Mark Benson wrote: Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferencing... ; Make as many of these contexts as you have seperate conference bridges ; change conferencename in each [conf-conferencename] exten = join,1,System(/opt/asterisk/bin/conference-announce conferencename in) exten = join,2,Conference(conferencename/S/1) exten = h,1,System(/opt/asterisk/bin/conference-announce conferencename out) [confhelper] ; make one of these extensions per seperate conference bridge exten = conf-conferencename,1,Conference(conferencename/S/1) exten = in,1,Answer() ; if I use Playback here instead of BackGround, asterisk crashes exten = in,2,BackGround(conf-announce) exten = in,3,ResponseTimeout(5) exten = in,4,Hangup() exten = out,1,Answer() exten = out,2,BackGround(conf-leave) exten = out,3,ResponseTimeout(5) exten = out,4,Hangup() how do I setup up app_conference to respond to an extention? Just a real simple example to get me started would be appreciated... I've tried a few things along the lines of the example meetme extention ie exten = 901,1,app_conference(901||1234) or exten = 901,1,cmd_conference(901||1234) But I guess its expecting too much to think that this would fireup app_conference Thanks in advance for any help. Cheers, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Caller ID name not showing.
On 15:54:12 June 29, 2005 Armin Schindler [EMAIL PROTECTED] wrote: On Wed, 29 Jun 2005, louis g wrote: I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4 port Eicon Diva card. All works fine, but i'd like calls from the PBX to Asterisk to show the Caller ID name and not just the number. I know this information is being presented by looking through the ISDN trace for the Eicon Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is '605'. Can anyone point me in the right direction to get this sorted?. It's works with X100P cards :) What 'name' do you mean? Is it a subaddress? Please paste an example for that Eicon card trace where you see that name. His PBX probably transmits the name per UUS1. zaphfc supports this also. I have a zaphfc card as internal ISDN and connected a Siemens ISDN DECT phone to it. Now, on incoming calls, the Siemens shows the CallerIDName as set by Asterisk in the display. zaphfc also supports SendText... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to fetch a call not in the same callgroup
Hi, the situation: A call rings at extension 123. My own extension is not in the same call- or pickupgroup for that extension. Is there a way to route the ringing extension 123 to my phone? Thanks, Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM card and voicemail volume
-Original Message- From: Steve Prior [mailto:[EMAIL PROTECTED] Here is the text of the last 2 bug comments by MikeJ (who I would assume closed the bug). text snipped I think there are three issues here: 1. The bug was originally filed as a feature request for a feature that would have been a work-around, at best. The actual source of the problem wasn't narrowed down until later. It probably should have been filed as a bug report, instead. Unfortunately, I fear that trying to file one now will probably just result in it being marked as a duplicate of the closed feature request. 2. I believe there are quite possibly two seperate bugs conflated in that one item. There's the recording format problem (compressed formats are at -6 or -10 dB compared to uncompressed) and possibly also a TDM-specific recording volume problem. 3. The people who are affected are not the people who are capable of fixing it, and the people who are capable of fixing it are apparently not affected enough by it to care. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] timeout on incoming PRI call
hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) so i assumed that the timeout was caused because asterisk didn't know if the number is complete or if further digits are sent, so i now replaced this config with a realtime config which lists each number individually. even when using this realtimeconfig (which includes only 'full' numbers - no wildcards, etc.) it seems that asterisk does the db-lookup after the timeout - so the delay is still there, although the dialed number is distinct. any suggestions about the cause of this problem / how to solve it? cu /gst signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration
http://www.gnugk.org/compiling-gnugk.html Also, the reqs for the included 323 channel and gnugk differ on versions. I have unreliably gotten them both to run on the same box with 100% reliability. Outbound calls transcoded from SIP - 323 - Gnugk - CCM - MGCP - PRI get dropped from DRQ after 2-4 seconds.. The README in the included channels/h323/README file will give you versions for openh323 and owlib that do not match any known working gnugk combo. Plus some applied patches from Janus. There is the new ooh323 channel driver out too (look on voip-info.org for info). I have not tried it as of yet, but it does not require openh323 and pwlib.. That combo and gnugk on same box may work well?? It is relatively new though. -Greg On Wed, 2005-06-29 at 14:28 +0100, Barney Sowood wrote: On Sat, Jun 25, 2005 at 07:58:24PM -0500, Greg Oliver wrote: That works well. You may also want to make sure your compatibility matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more issues than I care to talk about. The GNUGk web site has the best matrix to follow.. Do you have a specific URL, the only thing I can find is http://www.gnugk.org/interoperability.html, which doesn't sound exactly like what you're talking about. Thanks, Barney. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Lucent TNT echo
No I do not hear any clicking sound. Some calls come in perfect, and others come in with some echo and sometimes artifacts, which I think might be caused by jitter. Also it is mostly inbound calls that I have the problem with. If you didn't have any echo, just clicking, would you possibly still have a configuration that you could post so I can compare it with mine. I pretty much just followed the wiki for my config. Thanks, Jeremiah Date: Tue, 28 Jun 2005 15:56:24 -0700 From: Carlos [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk with Lucent TNT echo To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hey jeremiah, Do you hear a click click click sound I remember getting that with the licent tnt with the asterisk server main reason we stopped using the tnt. Carlos Alcantar Race Technologies, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.246.8900 F: 650.246.8901 E: carlos at race.com -Original Message- From: Jeremiah Millay [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 28, 2005 2:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with Lucent TNT echo I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3 calls I get is a bad one. Does anyone on the list run asterisk with this configuration? Does anyone have any tips to solve this issue? I've tried modifying the gains at the lucent, as well as turn off and on jitter buffers on asterisk. Tweaking these seems to help but I'm looking for something more solid. Any help would be appreciated. Regards, Jeremiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1
What does zttool say? Do you have any IRQ issues or anything? -- Tom On 6/29/05, Michael Blood [EMAIL PROTECTED] wrote: I receive this error on the asterisk console and it is pretty much ALWAYS coming up. Sometimes there will be a break where it does not display. We had our PRI provider test the lines and they claim that there is no signalling problem. It doesn't matter if there are no calls or if there are 10 calls in progress the error is still displayed. I also get an annoying popping or clicking sound but that doesn't always correspond with this error coming up so it is likely a separate issue. I have loaded all modules by hand like below as someone suggested in a search for HDLC errors on the list. insmod zaptel insmod wct1xxp Unfortunately it did not help Has anyone run into this in the past? Michael ;zapata.conf switchtype=national context=incoming_eli_pri_1 signalling=pri_cpe group=1 channel = 1-11 bchan=1-11 dchan=24 ;zaptel.conf span=1,1,0,esf,b8zs bchan=1-11 dchan=24 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:08 NOTICE[3094]: chan_zap.c:7394
Re: [Asterisk-Users] Trying to get *8 call pickup to work
I have been unable to get it to pickup sip-sip calls but if an incoming zap rings I can hit *8# and it works. My config is the same as yours: zapata has callgroup = 1 and in sip.conf I have pickupgroup = 1 I'm also using Grandstreams. t o n y On 6/28/05, Robert Woodcock [EMAIL PROTECTED] wrote: I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]: chan_sip.c:7402 handle_request: Nothing to pick up I'd appreciate hearing from anyone that has this working. Here's my sip.conf, features.conf, and zapata.conf: # zapata.conf sed 's/;.*//g' | grep -v ^$ [trunkgroups] [channels] context=default switchtype=national signalling=em_w rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived callprogress=yes musiconhold=default channel = 1-24 # features.conf sed 's/;.*//g' | grep -v ^$ [general] parkext = 700 parkpos = 701-720 context = parkedcalls pickupexten = *8 # sip.conf sed 's/;.*//g' | grep -v ^$ | grep -v '^[ ]' | sed s/secret=.*/secret=donttell/g [general] context=default callgroup=1 pickupgroup=1 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 callgroup=1 pickupgroup=1 context=default nat=no canreinvite=yes dtmfmode=rfc2833 incominglimit=4 [1310] username=1310 secret=donttell type=friend host=dynamic callerid=Grandstream SIP 1310 [EMAIL PROTECTED] [i1310] username=i1310 secret=donttell type=friend host=dynamic callerid=Grandstream SIP 1310 [1311] username=1311 secret=donttell type=friend host=dynamic callerid=John Jacob Jingleheime 1311 [EMAIL PROTECTED] [1312] username=1312 secret=donttell type=friend host=dynamic callerid=Cisco 7960G Test 1312 [EMAIL PROTECTED] FWIW, I get identical behavior with callgroup=/pickupgroup= specified for each extension. Here's some sanitized verbose output with SIP debugging enabled: -- Starting simple switch on 'Zap/24-1' Jun 28 10:43:18 DEBUG[16774]: chan_sip.c:771 __sip_autodestruct: Auto destroying call 'a01052a-13c4-42c104ea-371e-1957' Destroying call 'a01052a-13c4-42c104ea-371e-1957' Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 3 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 2 on Zap/24-1 Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:1381 zt_enable_ec: Enabled echo cancellation on channel 24 -- Executing Macro(Zap/24-1, stdexten|1312|SIP/1312) in new stack -- Executing Dial(Zap/24-1, SIP/1312|20) in new stack Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing Call for 1312 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter: Call from user '1312' is 1 out of 0 We're at asterisk.server.ip.addr port 19630 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x1 (g723) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760 From: asterisk sip:[EMAIL PROTECTED];tag=as61d8a13d To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 28 Jun 2005 17:43:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 284 v=0 o=root 17450 17450 IN IP4 asterisk.server.ip.addr s=session c=IN IP4 asterisk.server.ip.addr t=0 0 m=audio 19630 RTP/AVP 0 8 4 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to called.phone.ip.addr:5061 -- Called 1312 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760 From: asterisk sip:[EMAIL PROTECTED];tag=as61d8a13d To: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] Date: Tue, 28 Jun 2005 17:43:20 GMT CSeq: 102 INVITE Server:
Re: [Asterisk-Users] audiocodes
On 6/29/05, Joe Murray [EMAIL PROTECTED] wrote: Is anyone on this list using and audiocodes FXO gateway? I have Asterisk(1.07 on OS X) setup and working fine, including SIP phones and IAX2 phones - I can make outbound calls just fine and receive inbound calls just fine. However, I can't seem to find the right series of DTMF settings on the AudioCodes to allow DTMF tones to be sent after an outbound call is connected(phone banking, long distance provider etc...) while still allow the client devices(phones) to access Asterisk voicemail. It seems I can either have the phones use inband DTMF and work with the Audiocodes PSTN's or outband and work with Asterisk, but not both? Any help/thoughts/experiences would be appreciated... -joe I think there is the ability to set the options on the more recent firmwares (4.4 series) to allow either/or for DTMF types on the MP-108s and Mediant 2000 devices. I don't know exactly what or where the settings are, but be sure you've got the most recent firmware you can. They are generally better. I am not exacly using the AudioCodes devices so much with Asterisk as with SER, so perhaps the settings I saw wouldn't help you out at all. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Caller ID name not showing.
Hi, if you are using the QSIG protocol for the interconnection between Asterisk and the PBX, I have maybe a solution. for the X100P you are using Zapata driver of asterisk. (with the switchtype QSIG right?) But for the eicon you use the capi module? Caller Name within QSIG is standardized as Calling Name Identification Presentation (CNIP). CNIP is implemented in libpri/Zapata but not in the capi of asterisk. that's because CNIP is not standardized in capi. But we are lucky: Eicon has made some hacks in his capi driver, so it's possible to use CNIP with Eicon-Capi. I am writing at the moment on the implementation of Eicon-capi-CNIP for asterisk. hopefully it will work... Chris zaArmin Schindler wrote: On Wed, 29 Jun 2005, louis g wrote: I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4 port Eicon Diva card. All works fine, but i'd like calls from the PBX to Asterisk to show the Caller ID name and not just the number. I know this information is being presented by looking through the ISDN trace for the Eicon Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is '605'. Can anyone point me in the right direction to get this sorted?. It's works with X100P cards :) What 'name' do you mean? Is it a subaddress? Please paste an example for that Eicon card trace where you see that name. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth?
I have indeed already done so - I use G729 quite a bit since I travel alot in adverse conditions. Interesting thing is, I can get less choppy audio sometimes from my Vonage device using (what I suspect to be) Ulaw, while either ulaw or G729 will still give choppy response at that moment from my Linksys Paul - Original Message - From: Marcel van Kaam, Fonetica [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 12:28 AM Subject: RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth? You can set, in the linksys, the codec G729 for your line. In the Linksys also set only to use that codec. This can be done at the admin page of the line you use in the linksys. Also do that in the asterisk for your device. First buy the license from Digium. Then you will use less bandwidth and have a better sound upstream. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 1:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth? Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to the same hotel, I can get reliable connectivity. Assuming the hotel isn't helping me on the QOS front, and the Hotel's connectivity is the last word, then my Vonage ATA should be choppy, as well, no? This is what leads me to think I can do some tweaking later, Paul - Original Message - From: Greg Oliver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:17 PM Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth? Nothing you can do on this one.. Without the provider accepting your QoS settings, you are at their mercy. And yes, you are correct, most multi-tenant dwellings use xDSL for their connectivity due to it's price, and the upstream is usually less bandwidth than the downstream.. -Greg On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote: So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] timeout on incoming PRI call
I am not sure about E1 but it _should_ be the same. The Dialed Number is usually transferred in 'a whole block' as the Telco passing the call to you has already routed that call to you. What type of switch are you connected to?? Could your switch be expecting a ACK of some sort from *?? Have you turned on debugging? (pri debug span 1). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Günther Starnberger Sent: Wednesday, June 29, 2005 10:13 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] timeout on incoming PRI call hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) so i assumed that the timeout was caused because asterisk didn't know if the number is complete or if further digits are sent, so i now replaced this config with a realtime config which lists each number individually. even when using this realtimeconfig (which includes only 'full' numbers - no wildcards, etc.) it seems that asterisk does the db-lookup after the timeout - so the delay is still there, although the dialed number is distinct. any suggestions about the cause of this problem / how to solve it? cu /gst ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with OR Logic in the GotoIf Statement
I am having some
trouble implementing OR login in the GotoIf statement. I have
followed the examples in the Wiki and I still am getting a syntax
error.
Essentially I want
to screen for CallerIDs set for "Anonymous" OR "Unknown Caller". If
either of these are true I want to send it to statement 3 which clears the
CallerID and proceeds to Privacy Manager.
I have also tried
removing and adding quotes to no avail. I am running the 6/7/2005 CVS
Head.
exten =5000,1,NoOp,${CALLERIDNAME}exten
=5000,2,GotoIf($[$["${CALLERIDNAME}" =
"Anonymous"] | $["${CALLERIDNAME}" = "Unknown Caller"]]?3:5)exten
=5000,3,SetCIDNum()exten
=5000,4,SetCIDName()exten
=5000,5,PrivacyManagerexten
=5000,6,GotoIfTime(19:00-7:00|*|*|*?afterhours,s,1)exten
=5000,7,agi,astcalleridexten
=5000,8,DIAL(SIP/5001)exten
=5000,9,Voicemail(u5001)exten
=5000,110,Hangup
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-2",
"Anonymous") in new stackJun 29 10:34:09 WARNING[3946]: ast_expr.y:486
ast_yyerror: ast_yyerror(): syntax error: syntax error; Input:(Anonymous =
"Anonymous")|(Anonymous = "Unknown
Caller")^^^
^ -- Executing GotoIf("IAX2/[EMAIL PROTECTED]:4569-2",
"0?3:5") in new stack -- Goto
(in-out,7326031000,5)
smime.p7s
Description: S/MIME cryptographic signature
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Re: [Asterisk-Users] ASTCC not billing
the database exists because that is where the cards\PINs are stored, without the card\PIN I can not make a call, so the database exists, the permissions issue also may not be valid because when I set a connection charge the connection charge is recorded as billcost, but the cost of the call is not added to the connection charge to make the total billcost.
So, it seems that the astcc.agi file I am using has a problem, perhaps you can send me your WORKING astcc.agi.
Bernard Cresencia [EMAIL PROTECTED] wrote:
If astccdb exists, go to the database configurationpage [Configure] and change the database name to thecorrect one. You may have to set up permissions onthis database if it wasn't set up before. If itdoesn't exist, use the 'Create Database' button tocreate a new one.--- Ade Agbero <[EMAIL PROTECTED]>wrote: The reason for the problem is clear below, the ASTERISKCDRDB database is being updated instead of the ASTCCDB database which holds the cdrs and BILLCOST. How can this problem be corrected??? 3 Query UPDATE cards SET used='0' WHERE number='58767059' 3 Query UPDATE cards SET inuse='0' WHERE number='58767059' 3 Query SELECT * FROM cards WHERE number='58767059' 3 Query UPDATE cards SET used='0' WHERE number='58767059'<
BR>
3 Query UPDATE cards SET inuse='0' WHERE number='58767059' 3 Quit 4 Connect [EMAIL PROTECTED] on asteriskcdrdb 4 Query INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-06-29 05:17:42','\"Mike\" 1234','1234','777','TEST','SIP/1234-f285','SIP/213.45.62.117-1732','Dial','SIP/213.45.62.117/19315461298|30|HL(6:6:3)',42,42,'ANSWERED',3,'')Juan Luis Moyano <[EMAIL PROTECTED]> wrote: Bernard Cresencia wrote: sorry, I meant my.cnf, not my.conf. Once logging is enabled, I would do tail -f /var/log/myslqd.log and watch as the database is being accessed during a call. I've done what Bernard suggested and this is
my output from mysql.log on a successful call to number 612 on FWD. I'd like to know if any of you see something wrong or rare. Thanks a lot. Time Id Command Argument 050629 1:02:02 1 Connect [EMAIL PROTECTED] on astcc 050629 1:02:04 1 Query SELECT * FROM cards WHERE number='21' 1 Query SELECT * FROM cards WHERE number='21' 1 Query SELECT * FROM cards WHERE number='21' 1 Query SELECT * FROM cards WHERE number='21' 1 Query UPDATE cards SET used='1801' WHERE number='21' 1 Query UPDATE cards SET inuse='1' WHERE number='21' 050629 1:02:10 1 Query SELECT * FROM routes WHERE '612' RLIKE pattern ORDER BY LENGTH(pattern) DESC 050629 1:02:25 1 Query SELECT * FROM cards WHERE number='21' 1 Query SELECT * FROM trunks WHERE name='FWD' 050629 1:02:37 1 Query INSERT INTO
cdrs(cardnum,callerid,callednum,trunk,disposition,billseconds,billcost,callstart) VALUES ('21', '\"Coco\" 21', '612', 'FWD', 'ANSWER', '9', '150', 'Wed Jun 29 01:02:37 2005') 1 Query UPDATE cards SET used='1951' WHERE number='21' 1 Query UPDATE cards SET inuse='0' WHERE number='21' 1 Query SELECT * FROM cards WHERE number='21' 1 Query UPDATE cards SET used='1951' WHERE number='21' 1 Query UPDATE cards SET inuse='0' WHERE number='21' 1 Quit -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Play an announcement to the CALLING party
Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So, to avoid the employees forgetting it, an automatic announcement should be played. Besides, same rules are applicable for calls that may be recorded for quality assurance issues. At least for premium rate calls, queues won't work as the customer would strongly dislike hearing an announcement about the rate while waiting for an agent. The a() option of the dial app only works for CALLED parties and when trying to use a macro with the m() option, the Playback also goes to the called party. Anyone any hints on that? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Lucent TNT echo
On Wed, 2005-06-29 at 09:18 -0500, Jeremiah Millay wrote: No I do not hear any clicking sound. Some calls come in perfect, and others come in with some echo and sometimes artifacts, which I think might be caused by jitter. Also it is mostly inbound calls that I have the problem with. If you didn't have any echo, just clicking, would you possibly still have a configuration that you could post so I can compare it with mine. I pretty much just followed the wiki for my config. Thanks, Jeremiah How about a modem card with old firmware or even a bad modemcard. Did you try to reseat all the cards? Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Shoutcast Music On Hold problems?
um do I paste the below info in to a file and name it something? this looks really odd. from what my screen reader is reading to me it looks like to be some sort of script file or something - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 5:55 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? bash-3.00# cat musiconhold.conf | more ; ; Music on hold class definitions ; [classes] ; Christian Rock.NET ;default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ;loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ; Cleft in the Rock Radio (TESTING) default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ bash-3.00# pwd /var/lib/asterisk/mohmp3-empty bash-3.00# ls -la total 8 drwxr-xr-x 2 root root 4096 Jun 15 15:21 . drwxr-xr-x 9 root root 4096 Jun 15 15:18 .. -rw-r--r-- 1 root root0 Jun 15 15:21 empty.mp3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, June 29, 2005 1:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Fw: [Asterisk-Users] Shoutcast Music On Hold problems? - Original Message - From: hank [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I am using [EMAIL PROTECTED] 1.0 my mp3 is called mp3 it has nothing before it it is 0 bytes does my mp3 of 0 bytes need to have a .mp3 or does it need to be called anything? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 11:52 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? Worked for me with a different stream... I ran into this same problem before - but it was my own fault for not RTM... Both the manual and ast install advised of verifying correct version of mpg123... I had wrong version and thus got no noise... If you follow the directions explicitly laid out on the wiki you should have no problems. I use christianrock.net's shoutcast stream Like this in musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Tuesday, June 28, 2005 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
RE: [Asterisk-Users] Teliax Problems
One might also conclude that during the outage the support people were focusing on getting the system back up and were not near phones. At least that is what I would bet on. Just a thought considering how most of the smaller ITSPs seem to work. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Malcolm Taylor Sent: Wednesday, June 29, 2005 6:41 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Teliax Problems It's up and running again now. I just found it a little disconcerting not to be unable to reach their support numbers during the outage. Malcolm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, June 29, 2005 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Teliax Problems An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. I am sure it's just a crashed server, wait an hour and let the support people deal with it. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax Problems
From email I just rec'd from Teliax: Wed. 6/29/05 3am-6am Service Outage on voip-co1 'This morning starting at approximately 3am we experienced an unexpected outage on proxy voip-co1. The outage was the result of a thread collision between the proxy and it\\\'s database cluster. During this time subscribers registered to voip-co1 would not have been able to make or receive calls. Due to the passive nature of the problem the redundant system did not take over until approximately 6am. Steps have been taken to insure that in the event of a future problem the backup will immediately begin to handle all calls.' At 06:47 AM 6/29/2005, you wrote: Try voip-co2.teliax.com to register with. And read my other letter I suppose. This domain is apparently working as of 4:30, but have had the same problem since 1:30 AM PDT. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Malcolm Taylor |Sent: Wednesday, June 29, 2005 4:14 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Teliax Problems | | | |I'm currently unable to register with Teliax's server via IAX2 |and can't reach them via either of their phone numbers. Their |website is up and I have logged a support incident. | |Is anyone else experiencing the same problems? Having been |caught up in the Broadvoice fiasco a couple of months back, |I'm hoping that Teliax is not going through the same sort of thing. | |Malcolm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM card and voicemail volume
On Wed, 2005-06-29 at 10:05 -0400, David Brodbeck wrote: [snip] 2. I believe there are quite possibly two seperate bugs conflated in that one item. There's the recording format problem (compressed formats are at -6 or -10 dB compared to uncompressed) and possibly also a TDM-specific recording volume problem. Did anyone ever confirm that this is not a problem with cards from other vendors? Or is the issue unrelated to the hardware? 3. The people who are affected are not the people who are capable of fixing it, and the people who are capable of fixing it are apparently not affected enough by it to care. I guess you can vote with your dollars and buy cards from another vendor, start a picket line at Digium's offices or join forces and put up a bounty to get it fixed. While you are at it you might as well include a bounty for a solution for the frame slips that spandsp is suffering from. Wouldn't surprise me if they both shared a dark cold slippery cave with Gollum hidden deep inside the code :) Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't bridge between h323 and sip calls
Hello, I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that comes with asterisk. I tried to place a call from h323 device into asterisk. in extensions.conf, I routed the call to my sip phone. The sip phone was already registered with asterisk. all the signaling looks ok to me. The sip phone rings when h323 call hits the asterisk box. But then the call is dropped. It appears that asterisk is trying to convert incoming g.729 codec to ulaw and it can't. I was assumed that g.729 will pass-thru to the phone. In fact, when an invite is sent bothg G729, G723 are codecs in SDP. However, when SIP phone answers, it only replies with g723 on 200OK. I am still unclear about that, but that's not really that important. I would like to find out why I can't bridge these two legs. below is the trace from the call. I am suspecting that a line below is the cause, but not sure why. Can someone help??? Jun 29 10:59:46 WARNING[8862]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make H323/ip$64.243.115.153:32971/11679 compatible with SIP/debit-9f37 -asterisk log-- -- Executing Dial(H323/ip$64.243.115.153:32971/11679, SIP/debit|20|rt) in new stack Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g729 to ulaw Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g729 to ulaw We're at 64.243.115.157 port 18192 Answering with capability 0x1 (g723) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting (NAT) to 69.115.205.168:4152: INVITE sip:[EMAIL PROTECTED]:4146 SIP/2.0 Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 29 Jun 2005 14:59:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 292 v=0 o=root 8862 8862 IN IP4 64.243.115.157 s=session c=IN IP4 64.243.115.157 t=0 0 m=audio 18192 RTP/AVP 4 0 8 18 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called debit Jun 29 10:59:41 WARNING[8862]: chan_h323.c:588 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g729 to slin Jun 29 10:59:41 WARNING[8862]: indications.c:99 playtones_alloc: Unable to set 'H323/ip$64.243.115.153:32971/11679' to signed linear format (write) voip*CLI -- SIP read from 69.115.205.168:4152: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.16 Warning: 399 69.115.205.168 detected NAT type is symmetric NAT Content-Length: 0 --- (9 headers 0 lines)--- voip*CLI -- SIP read from 69.115.205.168:4152: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146;tag=2cfc88182690d7d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.16 Warning: 399 69.115.205.168 detected NAT type is symmetric NAT Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/debit-9f37 is ringing voip*CLI -- SIP read from 69.115.205.168:4152: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146;tag=2cfc88182690d7d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.16 Warning: 399 69.115.205.168 detected NAT type is symmetric NAT Contact: sip:[EMAIL PROTECTED]:4146 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 213 v=0 o=debit 0 8000 IN IP4 69.115.205.168 s=SIP Call c=IN IP4 69.115.205.168 t=0 0 m=audio 4192 RTP/AVP 4 101 a=sendrecv a=rtpmap:4 G723/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 11 lines)--- Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 69.115.205.168:4192 Found description format G723 Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 29 10:59:46 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g723 to ulaw Jun
Re: [Asterisk-Users] Teliax Problems
I use them and I have another friend with them so far they are okay, support is awesome, not any outages thus far and have been with them for about 3 weeks, not sure if they support iax or not, they do allow biod, prices are good. hth - Original Message - From: Chris Coulthurst [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 4:48 AM Subject: RE: [Asterisk-Users] Teliax Problems Does anyone have anything +/- to say about TeleSIP? They appear to have local DIDs where I live and all comments on the wiki indicate they are reputable.. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Wednesday, June 29, 2005 5:22 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Teliax Problems | | | I'm currently unable to register with Teliax's server via IAX2 and | can't reach them via either of their phone numbers. Their |website is | up and I have logged a support incident. | | Is anyone else experiencing the same problems? Having been |caught up | in the Broadvoice fiasco a couple of months back, I'm hoping that | Teliax is not going through the same sort of thing. | |An ethereal trace indicates the IP address is active, but it |is not responding to iax packets (registration). So, either |their asterisk app has failed or they have folded their tent as well. | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Connecting PBX to Asterisk
I would appreciate if someone can help me figure out what could be the problem in receiving the digits from the telrad switch/pbx. When you dial from the telrad, do you see any information being generated on the asterisk CLI? You may have to increase verbosity of the console by starting with something like asterisk -vvr to see this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OrderlyQ installations?
What experience can be shared about installing and running the OrderlyQ application? I have a bunch of queues set up and want to start adding some additional apps and this one looked promising but I have very little java experience and it doesnt seem to be running properly. Jason Kawakami ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to get *8 call pickup to work
Go get app_intercept from www.pbxfreeware.org /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 29, 2005, at 9:16 AM, Tony Nichols wrote: I have been unable to get it to pickup sip-sip calls but if an incoming zap rings I can hit *8# and it works. My config is the same as yours: zapata has callgroup = 1 and in sip.conf I have pickupgroup = 1 I'm also using Grandstreams. t o n y On 6/28/05, Robert Woodcock [EMAIL PROTECTED] wrote: I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]: chan_sip.c:7402 handle_request: Nothing to pick up I'd appreciate hearing from anyone that has this working. Here's my sip.conf, features.conf, and zapata.conf: # zapata.conf sed 's/;.*//g' | grep -v ^$ [trunkgroups] [channels] context=default switchtype=national signalling=em_w rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived callprogress=yes musiconhold=default channel = 1-24 # features.conf sed 's/;.*//g' | grep -v ^$ [general] parkext = 700 parkpos = 701-720 context = parkedcalls pickupexten = *8 # sip.conf sed 's/;.*//g' | grep -v ^$ | grep -v '^[ ]' | sed s/ secret=.*/secret=donttell/g [general] context=default callgroup=1 pickupgroup=1 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 callgroup=1 pickupgroup=1 context=default nat=no canreinvite=yes dtmfmode=rfc2833 incominglimit=4 [1310] username=1310 secret=donttell type=friend host=dynamic callerid=Grandstream SIP 1310 [EMAIL PROTECTED] [i1310] username=i1310 secret=donttell type=friend host=dynamic callerid=Grandstream SIP 1310 [1311] username=1311 secret=donttell type=friend host=dynamic callerid=John Jacob Jingleheime 1311 [EMAIL PROTECTED] [1312] username=1312 secret=donttell type=friend host=dynamic callerid=Cisco 7960G Test 1312 [EMAIL PROTECTED] FWIW, I get identical behavior with callgroup=/pickupgroup= specified for each extension. Here's some sanitized verbose output with SIP debugging enabled: -- Starting simple switch on 'Zap/24-1' Jun 28 10:43:18 DEBUG[16774]: chan_sip.c:771 __sip_autodestruct: Auto destroying call 'a01052a-13c4-42c104ea-371e-1957' Destroying call 'a01052a-13c4-42c104ea-371e-1957' Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 3 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 2 on Zap/24-1 Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:1381 zt_enable_ec: Enabled echo cancellation on channel 24 -- Executing Macro(Zap/24-1, stdexten|1312|SIP/1312) in new stack -- Executing Dial(Zap/24-1, SIP/1312|20) in new stack Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing Call for 1312 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter: Call from user '1312' is 1 out of 0 We're at asterisk.server.ip.addr port 19630 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x1 (g723) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760 From: asterisk sip:[EMAIL PROTECTED];tag=as61d8a13d To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 28 Jun 2005 17:43:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 284 v=0 o=root 17450 17450 IN IP4 asterisk.server.ip.addr s=session c=IN IP4 asterisk.server.ip.addr t=0 0 m=audio 19630 RTP/AVP 0 8 4 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to called.phone.ip.addr:5061 -- Called 1312 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760 From: asterisk sip:[EMAIL PROTECTED];tag=as61d8a13d To: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL
Re: [Asterisk-Users] Equipment for small office setup
1 Master phone for a receptionist. Is there an easy way at the moment for one of these bigger phones (cisco or whatever) to view the status of the various lines etc? Some phone with an expansion board maybe? Steve, Flash Operators Panel is a very good tool for a receptionist if they have a PC screen at their desk. I think it would be easier to see who's doing what on the phones with that than any SIP hardphone I've seen. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Play an announcement to the CALLING party
Why not play the message BEFORE you call the Dail application. This would also give the caller a chance to terminiate the call by hanging up BEFORE your techs even get the call.. Hint: use the playback application -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Wednesday, June 29, 2005 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Play an announcement to the CALLING party Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So, to avoid the employees forgetting it, an automatic announcement should be played. Besides, same rules are applicable for calls that may be recorded for quality assurance issues. At least for premium rate calls, queues won't work as the customer would strongly dislike hearing an announcement about the rate while waiting for an agent. The a() option of the dial app only works for CALLED parties and when trying to use a macro with the m() option, the Playback also goes to the called party. Anyone any hints on that? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting Caller ID after Dial
Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with Connecting PBX to Asterisk
I have tried increasing it to about verbosity level 11. Even then no sign of digits coming in. My telrad technician also came in and checked everything and certified the telrad is sending the digits as he switched cable on the T1 card with another card (connected to Telco) and showed that the dialing out worked fine on the same card with the same settings. To give you an idea, we use 99 to pick up trunk from telco and dialout to pstn on telrad, and asterisk is configured to be picked up by dialing 88. so when we switched the cables we were able to make calls by dialing 88 validating that the card was configured fine and the digits are being sent out. Any thoughts do you think there could be something trivial that I have missed in the configuration on the asterisk side to let asterisk know that this T1 span can dial out (works) and also receive calls (does not work) ? -Original Message- From: qrss [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 29, 2005 11:40 AM To: asterisk-users@lists.digium.com Cc: Karthik Natarajan Subject: Re: [Asterisk-Users] Problem with Connecting PBX to Asterisk I would appreciate if someone can help me figure out what could be the problem in receiving the digits from the telrad switch/pbx. When you dial from the telrad, do you see any information being generated on the asterisk CLI? You may have to increase verbosity of the console by starting with something like asterisk -vvr to see this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with OR Logic in the GotoIf Statement
Hi! Have you tried
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} =
Anonymous]|$[${CALLERIDNAME} = Unknown Caller]]?3:5)
instead of
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
?
Giorgio
Keith O'Brien wrote:
I am having some trouble implementing OR login in the GotoIf
statement. I have followed the examples in the Wiki and I still am
getting a syntax error.
Essentially I want to screen for CallerIDs set for Anonymous OR
Unknown Caller. If either of these are true I want to send it to
statement 3 which clears the CallerID and proceeds to Privacy Manager.
I have also tried removing and adding quotes to no avail. I am
running the 6/7/2005 CVS Head.
exten = 5000,1,NoOp,${CALLERIDNAME}
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
exten = 5000,3,SetCIDNum()
exten = 5000,4,SetCIDName()
exten = 5000,5,PrivacyManager
exten = 5000,6,GotoIfTime(19:00-7:00|*|*|*?afterhours,s,1)
exten = 5000,7,agi,astcallerid
exten = 5000,8,DIAL(SIP/5001)
exten = 5000,9,Voicemail(u5001)
exten = 5000,110,Hangup
_-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-2
mailto:IAX2/[EMAIL PROTECTED]:4569-2, Anonymous) in new stack
Jun 29 10:34:09 WARNING[3946]: ast_expr.y:486 ast_yyerror:
ast_yyerror(): syntax error: syntax error; Input:
(Anonymous = Anonymous)|(Anonymous = Unknown Caller)
^^^
^
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569-2
mailto:IAX2/[EMAIL PROTECTED]:4569-2, 0?3:5) in new stack
-- Goto (in-out,7326031000,5)_
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Re: [Asterisk-Users] Problems with OR Logic in the GotoIf Statement
Keith O'Brien wrote:
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
One too many $s?
exten = 5000,2,GotoIf($[${CALLERIDNAME} =
Doug
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Re: [Asterisk-Users] Problems with OR Logic in the GotoIf Statement
Hi! Try
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} =
Anonymous]|$[${CALLERIDNAME} = Unknown Caller]]?3:5)
intead of
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
Deleting spaces before and after ANd or OR logic worked for me.
Giorgio
Keith O'Brien wrote:
I am having some trouble implementing OR login in the GotoIf
statement. I have followed the examples in the Wiki and I still am
getting a syntax error.
Essentially I want to screen for CallerIDs set for Anonymous OR
Unknown Caller. If either of these are true I want to send it to
statement 3 which clears the CallerID and proceeds to Privacy Manager.
I have also tried removing and adding quotes to no avail. I am
running the 6/7/2005 CVS Head.
exten = 5000,1,NoOp,${CALLERIDNAME}
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
exten = 5000,3,SetCIDNum()
exten = 5000,4,SetCIDName()
exten = 5000,5,PrivacyManager
exten = 5000,6,GotoIfTime(19:00-7:00|*|*|*?afterhours,s,1)
exten = 5000,7,agi,astcallerid
exten = 5000,8,DIAL(SIP/5001)
exten = 5000,9,Voicemail(u5001)
exten = 5000,110,Hangup
_-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-2
mailto:IAX2/[EMAIL PROTECTED]:4569-2, Anonymous) in new stack
Jun 29 10:34:09 WARNING[3946]: ast_expr.y:486 ast_yyerror:
ast_yyerror(): syntax error: syntax error; Input:
(Anonymous = Anonymous)|(Anonymous = Unknown Caller)
^^^
^
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569-2
mailto:IAX2/[EMAIL PROTECTED]:4569-2, 0?3:5) in new stack
-- Goto (in-out,7326031000,5)_
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RE: [Asterisk-Users] Problems with OR Logic in the GotoIf Statement
If you need a fast solution put two gotoif
statements in a row, one to check for the first condition, another to check for
the next, you can leave out the redirect If the condition is not matched so it
just goes to the next priority.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keith O'Brien
Sent: Wednesday, June 29, 2005
8:40 AM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Problems
with OR Logic in the GotoIf Statement
I am having some trouble implementing OR login in the
GotoIf statement. I have followed the examples in the Wiki and I
still am getting a syntax error.
Essentially I want to screen for CallerIDs set for
Anonymous OR Unknown Caller. If either of
these are true I want to send it to statement 3 which clears the CallerID and
proceeds to Privacy Manager.
I have also tried removing and adding quotes to no
avail. I am running the 6/7/2005 CVS Head.
exten =5000,1,NoOp,${CALLERIDNAME}
exten =5000,2,GotoIf($[$[${CALLERIDNAME} =
Anonymous] | $[${CALLERIDNAME} = Unknown
Caller]]?3:5)
exten =5000,3,SetCIDNum()
exten =5000,4,SetCIDName()
exten =5000,5,PrivacyManager
exten =5000,6,GotoIfTime(19:00-7:00|*|*|*?afterhours,s,1)
exten =5000,7,agi,astcallerid
exten =5000,8,DIAL(SIP/5001)
exten =5000,9,Voicemail(u5001)
exten =5000,110,Hangup
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-2, Anonymous) in new stack
Jun 29 10:34:09 WARNING[3946]: ast_expr.y:486 ast_yyerror: ast_yyerror():
syntax error: syntax error; Input:
(Anonymous = Anonymous)|(Anonymous = Unknown Caller)
^^^
^
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569-2, 0?3:5) in new stack
-- Goto (in-out,7326031000,5)
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RE: [Asterisk-Users] Teliax Problems
I am assuming that you mean Telasip? Don't expect to get any numbers ported over to them. I have never been able to get anyone on the phone. Can't say that I have had any technical issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Wednesday, June 29, 2005 4:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Teliax Problems Does anyone have anything +/- to say about TeleSIP? They appear to have local DIDs where I live and all comments on the wiki indicate they are reputable.. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Wednesday, June 29, 2005 5:22 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Teliax Problems | | | I'm currently unable to register with Teliax's server via IAX2 and | can't reach them via either of their phone numbers. Their |website is | up and I have logged a support incident. | | Is anyone else experiencing the same problems? Having been |caught up | in the Broadvoice fiasco a couple of months back, I'm hoping that | Teliax is not going through the same sort of thing. | |An ethereal trace indicates the IP address is active, but it |is not responding to iax packets (registration). So, either |their asterisk app has failed or they have folded their tent as well. | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Caller ID name not showing.
On Wed, 29 Jun 2005, Christian Händel wrote: Hi, if you are using the QSIG protocol for the interconnection between Asterisk and the PBX, I have maybe a solution. for the X100P you are using Zapata driver of asterisk. (with the switchtype QSIG right?) But for the eicon you use the capi module? Caller Name within QSIG is standardized as Calling Name Identification Presentation (CNIP). CNIP is implemented in libpri/Zapata but not in the capi of asterisk. that's because CNIP is not standardized in capi. But we are lucky: Eicon has made some hacks in his capi driver, so it's possible to use CNIP with Eicon-Capi. I am writing at the moment on the implementation of Eicon-capi-CNIP for asterisk. hopefully it will work... That would be great :-) Armin___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth?
I have my systems running on ulaw, alaw or GSM. No other codecs. Myself I even prefer the ulaw because of the quality. I will look tomorrow a little bit further in the Linksys as I have 2 of them here to test and so far I am very happy with them. I will play a bit around with the settings and let you know tomorrow or I founded some things to improve. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 16:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth? I have indeed already done so - I use G729 quite a bit since I travel alot in adverse conditions. Interesting thing is, I can get less choppy audio sometimes from my Vonage device using (what I suspect to be) Ulaw, while either ulaw or G729 will still give choppy response at that moment from my Linksys Paul - Original Message - From: Marcel van Kaam, Fonetica [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 12:28 AM Subject: RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth? You can set, in the linksys, the codec G729 for your line. In the Linksys also set only to use that codec. This can be done at the admin page of the line you use in the linksys. Also do that in the asterisk for your device. First buy the license from Digium. Then you will use less bandwidth and have a better sound upstream. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 1:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth? Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to the same hotel, I can get reliable connectivity. Assuming the hotel isn't helping me on the QOS front, and the Hotel's connectivity is the last word, then my Vonage ATA should be choppy, as well, no? This is what leads me to think I can do some tweaking later, Paul - Original Message - From: Greg Oliver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:17 PM Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth? Nothing you can do on this one.. Without the provider accepting your QoS settings, you are at their mercy. And yes, you are correct, most multi-tenant dwellings use xDSL for their connectivity due to it's price, and the upstream is usually less bandwidth than the downstream.. -Greg On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote: So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
RE: [Asterisk-Users] Teliax Problems
Yes I was just reading that TeleSIP and Telasip are often mistaken, and was just editing my dialplan for my mistakes! When you meen porting numbers, I assume you are talking about LNP? If so, not a problem for me anyway. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rick Baranowski |Sent: Wednesday, June 29, 2005 8:34 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Teliax Problems | | |I am assuming that you mean Telasip? | |Don't expect to get any numbers ported over to them. | |I have never been able to get anyone on the phone. | |Can't say that I have had any technical issues. | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Chris Coulthurst |Sent: Wednesday, June 29, 2005 4:49 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Teliax Problems | |Does anyone have anything +/- to say about TeleSIP? They |appear to have local DIDs where I live and all comments on the |wiki indicate they are reputable.. | |Chris Coulthurst |[EMAIL PROTECTED] | | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of ||Rich Adamson ||Sent: Wednesday, June 29, 2005 5:22 AM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] Teliax Problems || || || I'm currently unable to register with Teliax's server via IAX2 and || can't reach them via either of their phone numbers. Their ||website is || up and I have logged a support incident. || || Is anyone else experiencing the same problems? Having been ||caught up || in the Broadvoice fiasco a couple of months back, I'm hoping that || Teliax is not going through the same sort of thing. || ||An ethereal trace indicates the IP address is active, but it ||is not responding to iax packets (registration). So, either ||their asterisk app has failed or they have folded their tent as well. || || ||___ ||Asterisk-Users mailing list ||Asterisk-Users@lists.digium.com ||http://lists.digium.com/mailman/listinfo/asteri|sk-users ||To ||UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
The CallerID that is seen by others on calls originating from your PRI is set by your PRI provider; you have no control from Asterisk about this as it gets overridden by the provider. You must contact your carrier and ask them to set the CallerID for all PRI lines to the desired name/number. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 29, 2005, at 08:33, Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
Chee Foong Chiew wrote: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. Don't use Caller*ID for billing. Use account codes, which is supported pretty much everywhere in Asterisk. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax Problems
That would have been understandable, but their phone lines both gave 'number unavailable' tones. I suppose this was because their lines use their own service. Malcolm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, June 29, 2005 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Teliax Problems One might also conclude that during the outage the support people were focusing on getting the system back up and were not near phones. At least that is what I would bet on. Just a thought considering how most of the smaller ITSPs seem to work. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Malcolm Taylor Sent: Wednesday, June 29, 2005 6:41 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Teliax Problems It's up and running again now. I just found it a little disconcerting not to be unable to reach their support numbers during the outage. Malcolm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, June 29, 2005 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Teliax Problems An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. I am sure it's just a crashed server, wait an hour and let the support people deal with it. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Good soft-phone on Linux
On Wed, 2005-06-29 at 10:40 +0200, Filippo Carone wrote: * Hamish Whittal ([EMAIL PROTECTED]) ha scritto: Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. it may be more of what you need but using asterisk with the OSS/Alsa module turns it in a very efficient client (it can run also without X installed ;) X-Lite for Linux has been working fairly well for me. http://www.xten.com/index.php?menu=productssmenu=download -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Red Hat Enterprise 3.0 issue
Hey federico, I have it working in a rhe 3.0 start asterisk in debug and see what it spits out probably a config issue. /usr/sbin/asterisk -vv -g -dd -c Carlos Alcantar Race Technologies, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.246.8900 F: 650.246.8901 E: carlos at race.com -Original Message- From: Federico Alves [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 28, 2005 7:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Red Hat Enterprise 3.0 issue I use this code: cd /usr/src/asterisk make config but the automatic startup for Red Hat does not work. My Red Hat is the 3.0 update 4. Has anybody made this work in the licensed version of Red Hat? It says Asterisk ended with exit status 0 Asterisk shutdown normally Any ideas? Federico Alves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OrderlyQ installations?
Jason Kawakami wrote: What experience can be shared about installing and running the OrderlyQ application? I have a bunch of queues set up and want to start adding some additional apps and this one looked promising but I have very little java experience and it doesn’t seem to be running properly. I don't mean to hijack this thread since the OP specifically mentioned OrderlyQ but - ICD (Intelligent Call Distributor) looks like it adds some sophistication to Asterisk ACD functionality and provides a flexible framework for customization: http://icd.sourceforge.net/tiki/tiki-index.php I'd be interested in hearing about ICD - specifically skills-based call routing, if anyone has done it. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC not billing
How does Asterisk calculate "BILLCOST", it appears the program for calculating BILLCOST may be wrong. Wherecan I locatethe program\file.Juan Luis Moyano [EMAIL PROTECTED] wrote: Has anyone noticed that the primary key in the cdrs table is cardnum? soit won't record more than the first call made by different cards.Perhaps I'm not understanding the purpose of de cdrs table. Maybe onesolution is to add an auto_increment uniqueid field like in theasteriskcdrdb cdr table. Can anyone point me in the right direction onthis one?-- Juan Luis Moyano[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with voicemail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Good soft-phone on Linux
Seth Remington wrote: On Wed, 2005-06-29 at 10:40 +0200, Filippo Carone wrote: * Hamish Whittal ([EMAIL PROTECTED]) ha scritto: Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. it may be more of what you need but using asterisk with the OSS/Alsa module turns it in a very efficient client (it can run also without X installed ;) X-Lite for Linux has been working fairly well for me. http://www.xten.com/index.php?menu=productssmenu=download -Seth For me, kiax (for KDE, with native iax-Support) runs fine. Hth Oliver ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma and quad card hang up problems
needhelp trying to figure out
why calls hang when using multple ports on Sangoma card.
we have 1 quad card with 3 T1 ports
configured, Port1 acts as connection to teleco (to our T1 PRI)
port 2 connects second system and
routes calls to port1
port 3 is Asterisk
pbx
calls all go in and out properly but
sometimes we get a call hang on when both sides hangup. this causes all calls to
fail until we restart * with restart now cmd. Which taks approx 10-20 seconds to
complete.
see log files form a call with show
channels at bottom.
This was an incoming call that was
answered and completed
== Spawn extension (pri-g3,
17087492476, 1) exited non-zero on 'Zap/47-1'
-- Hungup
'Zap/47-1'
-- Executing
Dial("Zap/1-1", "Zap/G3/7732997763|120") in new
stack
-- Called
G3/7732997763
-- Accepting call
from '7082975266' to '7732997763' on channel 0/1, span
1
-- Zap/47-1 is
ringing
-- Zap/47-1
answered Zap/1-1
-- Attempting
native bridge of Zap/1-1 and Zap/47-1
-- Hungup
'Zap/47-1'
== Spawn extension (default,
7732997763, 1) exited non-zero on 'Zap/1-1'
-- Call accepted
by 64.4.200.98 (format unknown)
-- Channel 0/1, span 1 got hangup
-- Channel 0/1, span 1 got hangup
Why 2
of the items in red?
Notice
we got both hang up request but below is what show channels states this is
after we both hung up on call.
NPS-816-Bwyn-Sw1*CLI show
channels
Channel
(Context Extension Pri ) State
Appl.
Data
Zap/1-1
(default 7732997763 1
) Up
(None)
(None)
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Re: [Asterisk-Users] How do you handle NAT?
Here is my experience in this area. Using asterisk on public IP no nat, and no firewall. Polycom and Sipura clients inside NAT. The sipura seems to be much more stable with almost everything, in terms of asterisk being able to connect to it. I'm not using qualify in sip.conf, but enabled them on the sipura. The polycoms however is a different story. For some reason, the polycoms allow asterisk to negotiate a new port so the port forwarding rules don't really help much, even though I set the polycoms to have different ports, and configured sip.conf with those ports. Because of that (or so I think) they become unreachable every 2-3 minutes until they reregister. Qualify (in sip.conf, since the polycoms don't have such a setting) helps a lot, but doesn't get rid of the problem. I even tried setting the register to 150 seconds but still it didn't help. It's actually set to this in sip.conf: maxexpirey=300 defaultexpirey=90 The main thing I'm trying to do now, that I think might help, is getting asterisk to stick to the port number, and not renegotiate a new port (the sipura works that way). Since I have port forwarding rules on the setup port for each phone, I believe it will get rid of most of the trouble. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dtmfmode=inband still broken in *-1.0.7
asterisk 1.0.7-r1 stable just came out on Gentoo but dtmfmode=inband is still broken. The work around is to use rfc2833 Was it fixed in ver. 1.0.8 ? -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth?
You may also want to do some packet captures when you experience the problem for both the Linksys and the Vonage ATA to see what they do differently.. -Greg On Wed, 2005-06-29 at 17:59 +0200, Marcel van Kaam, Fonetica wrote: I have my systems running on ulaw, alaw or GSM. No other codecs. Myself I even prefer the ulaw because of the quality. I will look tomorrow a little bit further in the Linksys as I have 2 of them here to test and so far I am very happy with them. I will play a bit around with the settings and let you know tomorrow or I founded some things to improve. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 16:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth? I have indeed already done so - I use G729 quite a bit since I travel alot in adverse conditions. Interesting thing is, I can get less choppy audio sometimes from my Vonage device using (what I suspect to be) Ulaw, while either ulaw or G729 will still give choppy response at that moment from my Linksys Paul - Original Message - From: Marcel van Kaam, Fonetica [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 12:28 AM Subject: RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth? You can set, in the linksys, the codec G729 for your line. In the Linksys also set only to use that codec. This can be done at the admin page of the line you use in the linksys. Also do that in the asterisk for your device. First buy the license from Digium. Then you will use less bandwidth and have a better sound upstream. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 1:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth? Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to the same hotel, I can get reliable connectivity. Assuming the hotel isn't helping me on the QOS front, and the Hotel's connectivity is the last word, then my Vonage ATA should be choppy, as well, no? This is what leads me to think I can do some tweaking later, Paul - Original Message - From: Greg Oliver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:17 PM Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth? Nothing you can do on this one.. Without the provider accepting your QoS settings, you are at their mercy. And yes, you are correct, most multi-tenant dwellings use xDSL for their connectivity due to it's price, and the upstream is usually less bandwidth than the downstream.. -Greg On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote: So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
RE: [Asterisk-Users] Asterisk with Lucent TNT echo
The Lucent has fairly new cards in it. We just had firmware upgraded to 11.0.2 I believe. I'm thinking it is a configuration issue either in asterisk or the lucent. Just wondering if anyone is running SIP between asterisk and a Lucent TNT successfully without any echo or problems of that nature. Thanks, Jeremiah Date: Wed, 29 Jun 2005 16:58:13 +0200 From: Patrick [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk with Lucent TNT echo To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Wed, 2005-06-29 at 09:18 -0500, Jeremiah Millay wrote: No I do not hear any clicking sound. Some calls come in perfect, and others come in with some echo and sometimes artifacts, which I think might be caused by jitter. Also it is mostly inbound calls that I have the problem with. If you didn't have any echo, just clicking, would you possibly still have a configuration that you could post so I can compare it with mine. I pretty much just followed the wiki for my config. Thanks, Jeremiah How about a modem card with old firmware or even a bad modemcard. Did you try to reseat all the cards? Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
Bryce Chidester wrote:
The CallerID that is seen by others on calls originating from your
PRI is set by your PRI provider; you have no control from Asterisk
about this as it gets overridden by the provider. You must contact
your carrier and ask them to set the CallerID for all PRI lines to
the desired name/number.
Regards,
Bryce Chidester
There must be different types of PRI lines because I was really shocked
when I started testing my Asterisk box on my PRI and the people
receiving the calls were flipping out because their caller id display
was showing my 3 digit SIP extensions. I wanted all outbound calls to
have the same callerid so I did it like this:
extensions.conf
[trunklocal]
exten = _6NX,1,SetCallerID(youroutboundnumber)
exten = _6NX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _6NX,3,Congestion
There was also a callerid option in zapata.conf, but I don't think it
had any affect for me. Good luck!!
Mark
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RE: [Asterisk-Users] Setting Caller ID after Dial
Ummm are you sure about this... I've seen people outpulse on PRI before It's dependent on the carrier - was my understanding. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryce Chidester Sent: Wednesday, June 29, 2005 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Setting Caller ID after Dial The CallerID that is seen by others on calls originating from your PRI is set by your PRI provider; you have no control from Asterisk about this as it gets overridden by the provider. You must contact your carrier and ask them to set the CallerID for all PRI lines to the desired name/number. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 29, 2005, at 08:33, Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF I forgot in my last post to mention that I use Postgres for my CDR, and the SIP extension can be pulled from the channel column. That way, the callerid is still the way it appeared when the calls were placed. I just strip everything from the '-' to the right and it's worked great for me! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
