[Asterisk-Users] IAX compatible phones
Hello, I would like to know which phones are IAX compatible. Thank-you Marios Moutzouris -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] X101P register map data please?
Hi Newbie, or would you prefer to be called VoIP(y)? :-) Thanks for the advice, It's great to hear from somebody that has suffered in the same way :-) I've cc'd in the dev and user lists mostly so that others looking for the same issue (FXO PCI Master Abort) can find some info! - hope you dont mind... On the card itself. I am assured by the vendor that they had the card up and running in a machine. Indeed, the vendor has taken one home, called me through it, and then given it to me... so I'm reasonably sure that under SOME situations, these cards work and I now have 2 of these cards! I'd be interested to know, are all the 3 cards you have had identical in terms of how they look? Do you still have them? Can we compare notes - (off list)? On the Zaptel driver... There are clearly inconsistencies in the driver, which I feel should be sorted out However, they are in code which people with working systems say is never reached. So.. yes, the driver should be cleaned up in order to handle the IRQ's better, but the question remains, why am I/you getting the Master Aborts in the first place... If the patch that I've done to the driver is the right thing to do, then maybe thats an answer for me/you/others. I still seem to have some problems, so I need to understand those first (see other post). At least the Master Abort doesn't bring the whole machine down. What I can't tell is why WE get the Master Aborts in the first place Speculation would be good! Any ideas? Cheers Mark. On 17 Aug 2005, at 07:16, VoIP Newbie wrote: Dear Mark, I got 3 X101P clone cards from 3 different vendors. One of them has the same problem like yours. Another one has echo issue. Only one from www.broad-tel.com works fine for me. You may want to contact the vendor and get one for yourself instead of modifying ZAPTEL software. Newbie On 8/16/05, Mark Burton [EMAIL PROTECTED] wrote: Hi, I've been trying to debug the problem with the X101P giving FXO PCI Master Aborts... I'm doing this blind, and I really need some info on the X101P's register map - or best of all, the conditions under which it can generate an IRQ with a mask of 0x10. I have so far set up the mask for the IRQ's in the interrupt handler (so the poor thing doesn't keep getting them)[as per previous post], then patched ztcfg so it actually starts the watchdog (which is assumed by the driver, but in reality doesn't happen - of course it doesn't need to, because under normal conditions there is no need for the watchdog - I guess?) That much gives me a system which runs, hits a PCI Master abort (or at least an IRQ with a mask of 0x10), and then stops the dma, masks the IRQ... then the watchdog starts the dma again, unmasks the IRQ, at which point it gets another IRQ before the next watchdog beat so the watchdog can't help. I have tried being a bit more brutal with the activities in the watchdog routine I just caused myself some kernel panic's :-) Again, any help appreciated! Cheers Mark. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] IAX compatible phones
For example TEK SIP-IAX 323. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dr. Marios Moutzouris Sent: Wednesday, August 17, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAX compatible phones Hello, I would like to know which phones are IAX compatible. Thank-you Marios Moutzouris -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/8/2005 ___ 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] Voipbuster blocking Asterisk/IAX connections?
On 17 Aug 2005, at 02:26, Don Fanning wrote: I've surmized that it's Voipbuster having issues. Paid up another euro on the second account and it works fine. When their support gets better, I'll have them work on the other account. I've had similar flakyness with Voipbuster. Sometimes the call goes through a dream, next time I either get no authority found or invalid extension/context. For me it's 50/50 This seems odd.. I put it down to their free service ... [Though, what s worse, If Voipbuster fails, then voipjet fails too, in the same way, and that I REALLY dont understand! But I haven't got on that case to Voipjet yet - so i dont know what the problem is...] Cheers Mark. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, August 16, 2005 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? I added in a second account that does not have the 1 Euro deposit and it goes through. What would make things so different? (this time the number is to the NIST Atomic Clock) --- *CLI iax2 debug IAX2 Debugging Enabled -- Executing SetCallerID(SIP/100-d2c1, jfalcon) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569] VERSION : 2 CALLED NUMBER : 0013034997111 CALLING NAME: jfalcon LANGUAGE: en USERNAME: jfalcon FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185631973 -- Called [EMAIL PROTECTED]/0013034997111 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] AUTHMETHODS : 3 CHALLENGE : 188826810 USERNAME: jfalcon Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] FORMAT : 2 -- Call accepted by 213.61.187.146 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] -- Hungup 'IAX2/voipbuster/10' == No one is available to answer at this time -- Executing NoOp(SIP/100-d2c1, DIALSTATUS=NOANSWER) in new stack -- Executing NoOp(SIP/100-d2c1, HANGUPCAUSE=0) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 5 DCall: 0 [213.61.187.147:4569] VERSION : 2 CALLED NUMBER : 0013034997111 CALLING NAME: jfalcon LANGUAGE: en USERNAME: jfalcontwo FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185631979 -- Called [EMAIL PROTECTED]/0013034997111 Rx-Frame Retry[ No] --
[Asterisk-Users] Re: [Asterisk-Dev] X101P register map data please?
Hi Newbie, or would you prefer to be called VoIP(y)? :-) Thanks for the advice, It's great to hear from somebody that has suffered in the same way :-) I've cc'd in the dev and user lists mostly so that others looking for the same issue (FXO PCI Master Abort) can find some info! - hope you dont mind... On the card itself. I am assured by the vendor that they had the card up and running in a machine. Indeed, the vendor has taken one home, called me through it, and then given it to me... so I'm reasonably sure that under SOME situations, these cards work and I now have 2 of these cards! I'd be interested to know, are all the 3 cards you have had identical in terms of how they look? Do you still have them? Can we compare notes - (off list)? On the Zaptel driver... There are clearly inconsistencies in the driver, which I feel should be sorted out However, they are in code which people with working systems say is never reached. So.. yes, the driver should be cleaned up in order to handle the IRQ's better, but the question remains, why am I/you getting the Master Aborts in the first place... If the patch that I've done to the driver is the right thing to do, then maybe thats an answer for me/you/others. I still seem to have some problems, so I need to understand those first (see other post). At least the Master Abort doesn't bring the whole machine down. What I can't tell is why WE get the Master Aborts in the first place Speculation would be good! Any ideas? Cheers Mark. On 17 Aug 2005, at 07:16, VoIP Newbie wrote: Dear Mark, I got 3 X101P clone cards from 3 different vendors. One of them has the same problem like yours. Another one has echo issue. Only one from www.broad-tel.com works fine for me. You may want to contact the vendor and get one for yourself instead of modifying ZAPTEL software. Newbie On 8/16/05, Mark Burton [EMAIL PROTECTED] wrote: Hi, I've been trying to debug the problem with the X101P giving FXO PCI Master Aborts... I'm doing this blind, and I really need some info on the X101P's register map - or best of all, the conditions under which it can generate an IRQ with a mask of 0x10. I have so far set up the mask for the IRQ's in the interrupt handler (so the poor thing doesn't keep getting them)[as per previous post], then patched ztcfg so it actually starts the watchdog (which is assumed by the driver, but in reality doesn't happen - of course it doesn't need to, because under normal conditions there is no need for the watchdog - I guess?) That much gives me a system which runs, hits a PCI Master abort (or at least an IRQ with a mask of 0x10), and then stops the dma, masks the IRQ... then the watchdog starts the dma again, unmasks the IRQ, at which point it gets another IRQ before the next watchdog beat so the watchdog can't help. I have tried being a bit more brutal with the activities in the watchdog routine I just caused myself some kernel panic's :-) Again, any help appreciated! Cheers Mark. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] Voicemail file permissions
On Tue, Aug 16, 2005 at 02:40:36PM -0400, hugolivude wrote: I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I'd like to give my Asterisk users the option of cleaning up their voicemail mailbox from their Windows PCs. I set up Samba and added all the users with restricted access to their mailbox only, but here's the problem: The voicemail .wav files that Asterisk creates have root as both owner and group. Since the users do not have root privileges, they can't do much with the files. BTW I'm not sure why the voicemail .wav files have root as both owner and group because I followed the instructions for running Asterisk other than root (see http://www.voip-info.org/wiki-Asterisk+non-root). Which is a good thing regardless. Is there a way around this w/o giving everyone root privileges! Do you want to allow every user to delete another user's voicemail? If not, how do you sync voicemail users and samba users? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] TE410P + SPANDSP fax problem
Ma Zhiyong schrieb: ... Trace shows that the fax is received successfully. Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX(Zap/94-1, Hi, sorry, I don't know the solution to your problem, but I would like to know, how did you get that trace? I'm looking for a reliable way to determine, whether TxFax did send a fax completely. I also tried the option debug, but never saw such a trace. Which version of spandsp are you using? Roger. ___ 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 eyebeam - video phone
Thank you for your answer. I didn't register on the domain of the Eyebeam software, actually I don't understand how to do that! I bouught 5 eyebeam activation keys and I am trying with the first 2 of them On the Eyebeam side (both eyebeam), I only enabled the Basic H.263 codec, no other. If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the two video phone speak without any problem (but without any video) If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the first video phone call the second, the second answer and immediately the call ends. If Ilook at /var/log/asterisk/full, I see: Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi completed, returning 0 Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial(SIP/551-eac0, SIP/552|25|tr) in new stack Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL) Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0 Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0 Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552 Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0 Aug 17 08:37:06 VERBOSE[14731]: -- Called 552 Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102 Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0 Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2) to SIP/552-ff46(524288) Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make SIP/551-eac0 compatible with SIP/552-ff46 Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse counter It seems the problem documented in bug http://bugs.digium.com/bug_view_page.php?bug_id=0003709 but actually it is not exactly the same. moreover: is there any way to put the patch described in http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *) in asterisk 1.0.9 and not asterisk CVS HEAD ? Any help will be greatly appreciated. Andrea Carlos Alperin [EMAIL PROTECTED] om.netTo Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 16/08/2005 20.48 RE: [Asterisk-Users] problems with eyebeam - video phone Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi, I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I only use H.263 and SIP. (G.729) Now, the more important question is if you register on the domain on the Eyebeam software. I found that this was the full secret about this. Let me know your configuration on the Eyebeam side. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 11:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problems with eyebeam - video phone I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm
[Asterisk-Users] Can not dial more then 23 calls
We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers terminate to another PBX that we can monitor. We have dialed out using cell phones to this other PBX while the test is happening and it connects, meaning it has more then 23 active calls on it. If we place more then 23 calls then it seems to 'queue' the extra calls, though not all of the extra calls complete after we stop adding new calls. They seem to get stuck in a queue or lost. We will send 200 calls through the Asterisk server and all but about 20 do eventually complete. Those 20 or so are stuck as Asterisk thinks the channels are busy with the calls when in fact there are no 'real' calls on the server. We can send 30 calls through the LD or PRI and only 23 are actually connected at a time. We can send 30 calls to both LD and PRI at the same time and still only a mixture of 23 calls are actually active at one time. So our issue seems to be located in our Asterisk server. Is there a way to limit or throttle an Asterisk server so that it will not place more then 'x' calls? We need to be able to support 48 calls. Any ideas? ___ 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] Issue with DTMF Tones - Codec Issues
I took a look at the NEAX brochures available from NEC's website. I may be wrong but I don't think you could change the way dtmf tones are sent from the PBX, but you should be able to send them out of band (with RTP, as per RFC 2833) from the cisco to the asterisk box. Generally, out of band dtmf is always better (when available) and more reliable than inband dtmf. Bear in mind that certain phones, such as grandstreams, do not work well with rfc2833 dtmf relaying, but work well with dtmf sent in SIP INFO messages. cheers On 8/16/05, Aaron W [EMAIL PROTECTED] wrote: Thanks I give give that a try. One follow up question. If the call is coming in via the PSTN, and going through the NEAX (PBX) then to the Cisco, can I control the way the PBX sends the DTMF, or is the cisco some how able to split out the DTMF tones from everything else? I was assuming that becuase I am going through the PBX, the cisco would recieve the DTMF inband, and therefore it would have to send it out also as inband. Thanks again Aaron On 8/16/05, maka [EMAIL PROTECTED] wrote: just a suggestion, but why don't you try using RFC2833 dtmf relay between the cisco and the asterisk box. use dtmfmode=rfc2833 in sip.conf, and you can also set the dtmf mode per peer in sip.conf also, if you use inband dtmf, this would only work with u-law and a-law, and not g729. on the cisco, enter Router(config-dial-peer)# dtmf-relay rtp-nte in dial-peer configuration mode. I recently had problems with a cisco gw forwarding pstn dtmf digits to my asterisk box, and rfc2833(which is what rtp-nte stands for in cisco's terms) solved it successfully. cheers On 8/16/05, Aaron W [EMAIL PROTECTED] wrote: Topology: PSTN-T1 PRI-NEAX2400-T1 PRI-Cisco 3825-Ethernet- Asterisk VoIP server When I make a call to a VoIP user from the PSTN, the call gets routed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my phone. When I call that number again via the PSTN, and I am prompted to enter my extension number Asterisk never hears the dtmf tones. I have done some digging around, and my guess is that the issue relates to the codec being used messing up the tones. Am I on the right track? Is there a ideal way to handle this? what do others do? I have posted my sip.conf below. Thanks, Aaron [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls (default context has no routing for security purposes) ;dtmfmode=rfc2833 dtmfmode=inband srvlookup = yes disallow=all; Disallow all codecs ;allow=g729 ; Codecs that we allow (in order of preference) allow=ulaw ;allow=alaw allow=g729 ;allow=ulaw ;allow=all [3120] callerid=Aaron Walsh 3120 type=friend host=dynamic canreinvite=no qualify=yes nat=yes setvar=LDPREFIX=199 context=XXX secret=X [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 -- I'm sick and tired of being sick and tired... ___ 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 -- I'm sick and tired of being sick and tired... ___ 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] Voipbuster blocking Asterisk/IAX connections?
They're using the same hosted servers with different billin schemes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Burton Sent: Tuesday, August 16, 2005 11:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? On 17 Aug 2005, at 02:26, Don Fanning wrote: I've surmized that it's Voipbuster having issues. Paid up another euro on the second account and it works fine. When their support gets better, I'll have them work on the other account. I've had similar flakyness with Voipbuster. Sometimes the call goes through a dream, next time I either get no authority found or invalid extension/context. For me it's 50/50 This seems odd.. I put it down to their free service ... [Though, what s worse, If Voipbuster fails, then voipjet fails too, in the same way, and that I REALLY dont understand! But I haven't got on that case to Voipjet yet - so i dont know what the problem is...] Cheers Mark. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Fanning Sent: Tuesday, August 16, 2005 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? I added in a second account that does not have the 1 Euro deposit and it goes through. What would make things so different? (this time the number is to the NIST Atomic Clock) --- *CLI iax2 debug IAX2 Debugging Enabled -- Executing SetCallerID(SIP/100-d2c1, jfalcon) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569] VERSION : 2 CALLED NUMBER : 0013034997111 CALLING NAME: jfalcon LANGUAGE: en USERNAME: jfalcon FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 185631973 -- Called [EMAIL PROTECTED]/0013034997111 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] AUTHMETHODS : 3 CHALLENGE : 188826810 USERNAME: jfalcon Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] FORMAT : 2 -- Call accepted by 213.61.187.146 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569] -- Hungup 'IAX2/voipbuster/10' == No one is available to answer at this time -- Executing NoOp(SIP/100-d2c1, DIALSTATUS=NOANSWER) in new stack -- Executing NoOp(SIP/100-d2c1, HANGUPCAUSE=0) in new stack -- Executing Dial(SIP/100-d2c1, IAX2/[EMAIL PROTECTED]/0013034997111) in new stack Tx-Frame Retry[000] --
Re: [Asterisk-Users] florz patch for bristuff breaks compile on x86_64?
On Wed, 17 Aug 2005, Tzafrir Cohen wrote: On Wed, Aug 17, 2005 at 06:57:19AM +0200, Remco Barende wrote: After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). Then merge the fix into the bristuff patch if it has not been merged yet! That's what I did when I patched bristuff :) It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Latest bristuff is RC8n, BTW. What exactly is the florz patch? It seems to have been onchanged since January or so. I have never ever been able to keep a bristuffed box up for more than a few hours or 2 days at best without the florz patch. It seems that KPJ is trying various approaches to solve timing problems but I guess it's not stable yet. Florz fixes a lot of timing issues, reduces interrupt load and makes bristuff stable. You can find more info here: http://zaphfc.florz.dyndns.org/ Anyone managed to get bristuff with florz working on x86_64 arch? It is part of the debian packages and they are built on amd64 as well. http://packages.debian.org/zaptel http://packages.debian.org/libpri http://packages.debian.org/asterisk I would guess thet are without bristuff and/or florz? Bristuff compiles without florz, but zaphfc doesn't after applying florz. ___ 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] Nikotel issues
Hi!
I've read in the archives that there are problems concerning Nikotel calls
being disconnected after two minutes. I had the same problem yesterday. Is
there a fix? There was only a giving up statement after the last e-mail in
the archive, I'm about to do that too.
Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's
working):
[nikotel]
type=friend
host=calamar0.nikotel.com
username=user
secret=pass
fromuser=user
fromdomain=nikotel.com
qualify=yes
context=nikotel-incoming
insecure=very
canreinvite=no
promiscredir=yes
diallow=all
allow=alaw
allow=ulaw
allow=gsm
extension.conf:
[nikotel-incoming]
exten = 3740525,1,NoOp(Invoming call via nikotel-us)
exten = 3740525,2,Dial(IAX2/christophSIP/30${CONSOLE},30)
exten = 3740525,3,VoiceMail(u30)
exten = 3740525,4,Hangup
Thanks for any help,
Christoph
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Re: [Asterisk-Users] TxFax - RxFax on same machine hangs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 17 Aug 2005, Steve Underwood wrote: Roger Schreiter wrote: How can I enable asterisk to fax to itsself? Well, it won't be the normal operation, but when allowing clients to fax, it can happen by chance, that someone faxes to another user on the same machine without knowing it. Thanks for any hints! If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. I can confirm that this problem appears on a call through the PSTN. My setup is: TxFax - Asterisk - E1 - Asterisk (same box) -RxFax Asterisk version 1.0.9 and spandsp version 0.0.2pre18 on debian woody (3.0). I sent you an email about it with some debug information a week or so ago. If you need it again, or need some other info I'll be happy to provide it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDAui2ckvkFeO3ANARAuJPAKC00b+lEeHz+mOfb8J/zOF7+YAwggCeLFrG KGkJxLFGCeBY6foyDqC1xGM= =J6zk -END PGP 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] Can not dial more then 23 calls
On Tue, 2005-08-16 at 23:53 -0700, Pudenz, Duane wrote: We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers terminate to another PBX that we can monitor. We have dialed out using cell phones to this other PBX while the test is happening and it connects, meaning it has more then 23 active calls on it. Include more info like your extensions.conf zaptel.conf and zapata.conf (not the comments) and we might be able to offer some hints... Also, some output from the CLI might be helpful that would probably tell you why the calls are failing already Regards, Adam ___ 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 change RINGING style for internal calls
I'd like to have the ringing a caller hears to be more like a 'british' ring when I am calling an internal extension. The phones I'm calling already do this, now I'd like to find a way to make the same thing happen for the caller who waits... Any ideas? 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
[Asterisk-Users] Voicemail Retrival
Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos___ 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] 1-800 number
Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ 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] Voicemail Retrival
On Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote:
Hi,
Hi!
Any ideas??
Yes, I do it in the following way. In extension.conf add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then automatically gets directed to their
voicemail where they have some options.
I hope this helps,
Christoph
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RE: [Asterisk-Users] Voicemail Retrieval
Take
this as an example
[from-sip]
exten
= 2000,1,Dial(SIP/2000,20)
exten
= 2000,2,Voicemail(u2000)
exten
= 2000,102,Voicemail(b2000)
exten
= 2000,103,Hangup
exten
= 2001,1,Dial(SIP/2001,20)
exten
= 2001,2,Voicemail(u2001)
exten
= 2001,102,Voicemail(b2001)
exten
= 2001,103,Hangup
exten
= 2999,1,VoicemailMain(${CALLERIDNUM})
you then dial 2999 to retrieve
it.
Kun
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Sharadindu
MohantySent: Wednesday, August 17, 2005 4:30 PMTo:
asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Voicemail
Retrival
Hi,
I am very new to Asterisk. I wanted to know how to retrive
the Voicemails. I could see some voicemails assosiated with some
extensions.
Any ideas??
How much free photo storage do you get? Store your
holiday snaps for FREE with Yahoo! Photos. Get
Yahoo! Photos
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RE: [Asterisk-Users] 1-800 number
try bankone, their 1800 waiting is long -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christoph Eicke Sent: Wednesday, August 17, 2005 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 1-800 number Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ 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] 1-800 number
More info On 8/17/05 3:34 AM, Christoph Eicke [EMAIL PROTECTED] wrote: Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ 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] 1-800 number
On Wednesday 17 August 2005 10:45, Michael K. Rodriguez wrote: More info I don't quiet understand your mail ;-) Do you want more info from me? ___ 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] 8 FXS in Asterisk Server
Thanks for the hint, do you know where to buy it (cheap) and the price for it? Thanks, Roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie Sent: Wednesday, August 17, 2005 6:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 8 FXS in Asterisk Server Get a 8-port FXS gateway from www.broad-tel.com. That is the single box you need. On 8/16/05, Roland Zagler [EMAIL PROTECTED] wrote: Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in advance, Roland ___ 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] FW: Asterisk-panel
I am running asterisk at home but have a strange phenomena that is going on with my flash panel I am using two ips an internal and an external public ip address on my box. If I go to the page on my asterisk external ip address the displays the flash panel everything is fine, but on the internal ip wich used to display the page correctly all the names of my panel users is there but the panel just flashes red and green does anyone knows why or how I can fix it? ___ 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] Asterisk (multiple) + Ser
I have several Asterisk servers installed and one SER server which will act as a gateway to PSTN, en redirect server. I was thinking to implement it the following way: - Register all the * servers at SER (is this neccessary?) - this works via register=asterisk:[EMAIL PROTECTED] in sip.conf - Setup aliases in SER for the telephonenumbers to the appropiate * server: serctl alias add [EMAIL PROTECTED] [EMAIL PROTECTED] e-mailaddress This way, when one SIP phone behind a * server calls for example 016234567, the * server forwards the request to SER, SER looks up the alias en then forwards it to the destined * server. If a number cannot be handled, SER will forward it to the PSTN gateway. Now my problems: I'm a totaly newby on SER. I managed to get the * server register themselves with SER, and setup Aliases. However I cannot get ser.conf configured so that it does what i've explained before. Is anybody willing to help me out, if possible with a sample ser.conf? TIA, Ronald Voermans ___ 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] Automatic start with SuSe linux
Hi! I'm trying to start asterisk at boottime. Since SuSe has no rc.local like in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory (I assume it is like that if i want automated asterisk startup). Do you have any experience how this is implemented in SuSe, and if you have some useful script for starting asterisk, I would be very, i mean VERY pleased? Thank you all in advance! This mail sent through L-secure: http://www.l-secure.net/ ___ 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] zaphfc ptp did problems
Hi All, i have a HFC card running in ptp mode. I set overlapdial to yes and immediate to no in /etc/zaptel.conf. DID works, but the timeout for immediate=no is much to low. Calls from GSM or via Speed Dial work fine, but you hardly can dial the digits fast enough to reach the extension (im using 777, most of the time im placed in the s context, sometimes 7 or 77) Is there a way to increase the timeout or any other workaround? regards, Hari mfg, Harald Klein ___ 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] Voicemail Retrival
I did the same way but it is asking for some password and mailbox. I think mail box is extension no but what abt password?
Can i overide this procedure?
ThanksChristoph Eicke [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote: Hi,Hi! Any ideas??Yes, I do it in the following way. In extension.conf add this line:exten = ,1,VoiceMailMain(s${CALLERIDNUM})exten = ,2,Hangup()Here any extension can call and then automatically gets directed to their voicemail where they have some options.I hope this helps,Christoph___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-usersSharadindu Mohanty
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[Asterisk-Users] zaphfc ptp DID problems
Hi All, so i have a better description: DID does not work with match as you go dialing, all at once is ok.. have a nice time, Hari Am 17.8.2005 schrieb Harald Klein [EMAIL PROTECTED]: Hi All, i have a HFC card running in ptp mode. I set overlapdial to yes and immediate to no in /etc/zaptel.conf. DID works, but the timeout for immediate=no is much to low. Calls from GSM or via Speed Dial work fine, but you hardly can dial the digits fast enough to reach the extension (im using 777, most of the time im placed in the s context, sometimes 7 or 77) Is there a way to increase the timeout or any other workaround? regards, Hari mfg, Harald Klein) mfg, Harald Klein ___ 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] Voicemail Retrival
Hi,
This procedure will work under one condition --
your user names are same as your extension numbers. I have same problem. I was
giving phones alphanumeric user names, like "phone1".
When VoicemailMain is called with ${CALLERIDNUM},
it is actually called as VoiceMailMain("phone1"). As a result, voice mail is
asking for a mailbox number which is same as your extension number. (BTW, is
there a way to extract extension number rather than phone name?).
As I am experimenting with *, I will rename phones
to match their extensions.
Rudolf
- Original Message -
From:
Sharadindu Mohanty
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 17, 2005 8:32
PM
Subject: Re: [Asterisk-Users] Voicemail
Retrival
I did the same way but it is asking for some password and mailbox. I
think mail box is extension no but what abt password?
Can i overide this procedure?
ThanksChristoph Eicke [EMAIL PROTECTED] wrote:
On
Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote:
Hi,Hi! Any ideas??Yes, I do it in the following way. In
extension.conf add this line:exten =
,1,VoiceMailMain(s${CALLERIDNUM})exten =
,2,Hangup()Here any extension can call and then
automatically gets directed to their voicemail where they have some
options.I hope this
helps,Christoph___Asterisk-Users
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Re: [Asterisk-Users] Voicemail file permissions
Great idea, thanks! I'd never heard of externnotify. I shudder to think of how many other cool features I'm missing! I'll let u know how it goes. Cheers, Hugh On 8/16/05, Chris Coulthurst [EMAIL PROTECTED] wrote: My suggestion would be, use the externnotify=/usr/bin/myapp feature in voicemail.conf to chown the permissions to something else. Since they are root, asterisk should have no problem deleting and moving them around with less privileges. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: hugolivude [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Tuesday, August 16, 2005 11:40 AM Subject: [Asterisk-Users] Voicemail file permissions I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I'd like to give my Asterisk users the option of cleaning up their voicemail mailbox from their Windows PCs. I set up Samba and added all the users with restricted access to their mailbox only, but here's the problem: The voicemail .wav files that Asterisk creates have root as both owner and group. Since the users do not have root privileges, they can't do much with the files. BTW I'm not sure why the voicemail .wav files have root as both owner and group because I followed the instructions for running Asterisk other than root (see http://www.voip-info.org/wiki-Asterisk+non-root). Is there a way around this w/o giving everyone root privileges! Thanks, Hugh ___ 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] Voicemail file permissions
Is there a way around this w/o giving everyone root privileges! Do you want to allow every user to delete another user's voicemail? If not, how do you sync voicemail users and samba users? I want each user to see, read and write (delete) their own voicemail ONLY (i.e. a user shouldn't be able to listen to someone elses voicemails). I gave each user an account on the Asterisk box and limited their access to their mailbox folder only. Hugh ___ 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] 5 way calling?
I'd not bother with using the flash based 3 way calling. Instead I'd setup an account with an ITSP and make the outbound calls via IP, preferabbly via IAX2. That way to can reach out to as many people as your bandwidth allows. Simply. Conveniently. Add one IP based DID and you can let others call in to your conference via IP. I've been thinking about getting some IP DIDs for other reasons anyway, so thanks for the suggestion. There's a bandwidth issue however and this client is simply more comfortable keeping things on copper, especially con-calls. As I mentioned the client's paying for 3-way calling from Bell, so is there no way to take advantage of this and establish a three way call on a single FXO line through Asterisk? I've opened another thread on this issue as it's more fundamental than my original 5 way calling problem. Thanks, Hugh ___ 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 + chan_mISDN = undefined symbol: ast_pickup_call
Hi! Then I get compile-errors. Greets, Christian Johann Steinwendtner schrieb: Christian Wengel schrieb: Hi! I tried install-misdn.tgz from http://www.beronet.com/download/ , some minutes ago. Also I switched to an older kernel (2.6.8), but I get the same error. I think that I made the correct changes in the Makefiles, but I will attach them to this e-mail, maybe you see something wrong. Is there a change when you uncomment this flag ? # ASTERISK Version # If you are using a asterisk version above from stable (v1-0) # then comment the following line out (good luck) # #CFLAGS+=-DASTERISK_STABLE Best regards Hans ___ 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] Polycom 501 dialing problem
Are you referring to the sip.conf setting or something in the phone's config? Sip.conf already reflects rfc2833. Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Tuesday, August 16, 2005 8:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 dialing problem Sounds like you have a DTMF mode problem. Check that you are using RFC2833 for dtmf signaling. I had the same thing happen with my dialing of *98 to check voicemail..It would transpose it in to 9*8, as if the * was being some sort of a tab key. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Craig Bruenderman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, August 16, 2005 11:55 AM Subject: [Asterisk-Users] Polycom 501 dialing problem When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ 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] Echo cancellation again ...
I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Thanks for clearing this up. Alan ___ 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] X100P dial out problem
Hi all! I'm new to asterisk and I'm trying a simple config with: - Debian GNU/Linux (unstable) - last version of Asterisk - a X100P card I have a problem with dial out from a SIP software phone (XLITE) to a public number (ex. my mobile phone), asterisk start the call, but nothing happen... If I run ztmonitor 1 I can see the right RX level and if I try to make a call with an analog standard phone connected to the second plug of the X100P, I can see the RX level going UP and down normally, and I can also hear my voice during a call. Otherwise, when I try to dial out from XLITE, when I start the call the RX level go to 0 and I can only hear the numbers of che called number but I can hear nothing on RX and the line is locked until I remove the wcfxo kernel module; in Italy we must wait for a tone before starting the call Is there anyone here with an idea for my problem ? Thanks in advance. Piero Baudino ___ 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] Echo cancellation again ...
I have a PRI with quite a nasty echo problem that I cannot seem to get rid of. I've tried all of the echo cancellation settings and tweaked gains to hell and back but still get echo. I am convinced it can only be addressed by hardware echo cancellation but that's not an option unless I replace my TE110P. Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alan Bunch Sent: Wednesday, August 17, 2005 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Echo cancellation again ... I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Thanks for clearing this up. Alan ___ 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] Voicemail Retrieval
In addition to my previos e-mail.
'callerid' filed in sip.conf or iax.conf
(depends where user is defined) must be set to"
callerid "User Name" EXT
Where EXT is a number that will be picked up by
VoiceMailMain and will be used as a mailbox number.
Rudolf
- Original Message -
From:
Wei Kun
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, August 17, 2005 6:37
PM
Subject: RE: [Asterisk-Users] Voicemail
Retrieval
Take
this as an example
[from-sip]
exten
= 2000,1,Dial(SIP/2000,20)
exten
= 2000,2,Voicemail(u2000)
exten
= 2000,102,Voicemail(b2000)
exten
= 2000,103,Hangup
exten
= 2001,1,Dial(SIP/2001,20)
exten
= 2001,2,Voicemail(u2001)
exten
= 2001,102,Voicemail(b2001)
exten
= 2001,103,Hangup
exten
= 2999,1,VoicemailMain(${CALLERIDNUM})
you then dial 2999 to retrieve
it.
Kun
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
Sharadindu MohantySent: Wednesday, August 17, 2005 4:30
PMTo: asterisk-Users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Retrival
Hi,
I am very new to Asterisk. I wanted to know how to retrive
the Voicemails. I could see some voicemails assosiated with some
extensions.
Any ideas??
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Re: [Asterisk-Users] Echo cancellation again ...
Alan Bunch wrote: I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Alan, I have experienced echo on our PRI with EC turned off. Granted, it was Asterisk server 1 connecting via IAX to server 2, connecting via a PRI to call my cell phone. Turning on EC removed this echo. 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] TxFax - RxFax on same machine hangs
Bartek Kania wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 17 Aug 2005, Steve Underwood wrote: Roger Schreiter wrote: How can I enable asterisk to fax to itsself? Well, it won't be the normal operation, but when allowing clients to fax, it can happen by chance, that someone faxes to another user on the same machine without knowing it. Thanks for any hints! If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. I can confirm that this problem appears on a call through the PSTN. My setup is: TxFax - Asterisk - E1 - Asterisk (same box) -RxFax Asterisk version 1.0.9 and spandsp version 0.0.2pre18 on debian woody (3.0). I sent you an email about it with some debug information a week or so ago. If you need it again, or need some other info I'll be happy to provide it. Did you put txfax in caller mode? Steve ___ 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] iaxcomm huge latency
Hello, I use iaxcomm-latest from the iaxclient.sf.net page (binary release) on linux, also tried Mac OS X version with the same result and Asterisk 1.0.9 from Debian. Iaxcomm has a huge latency -- tens of seconds, constantly changing over time. It was run on two different machines, always to a SIP phone (which otherwise works correctly even with VoipBuster, which also uses IAX with no latency and other SIP phones). Is it a known bug? Juraj. ___ 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] XORCOM RAPID Asterisk - Suggestions?
Hey Guys, Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now its fine as currently it is in testing stage with 3-4 users. Any Ideas??? ThanksSharadindu Mohanty How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos___ 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 astcc-config.conf card length question
I've done this without any problems. I changed from 10 digits to 11 digits and I'm still able to use all of the cards. --- Nate Kapi [EMAIL PROTECTED] wrote: I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks ___ 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] Echo cancellation again ...
I have experienced pretty nasty echo on my PRI w/TE110P. The echo was only coming from other POTS lines, because cell phones already have echo cancellation, and other PBX's had the same. I resolved the problem by turning on the AGGRESSIVE option and it works fine now, and we haven't noticed a severe degradation in sound quality - most of my operators were just happy the echo was gone :) -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/17/05, Doug Lytle [EMAIL PROTECTED] wrote: Alan Bunch wrote: I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Alan, I have experienced echo on our PRI with EC turned off. Granted, it was Asterisk server 1 connecting via IAX to server 2, connecting via a PRI to call my cell phone. Turning on EC removed this echo. 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 -- Tom ___ 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] Automatic start with SuSe linux
You could just add the line asterisk to /etc/init.d/boot.local Angus - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 17, 2005 11:27 AM Subject: [Asterisk-Users] Automatic start with SuSe linux Hi! I'm trying to start asterisk at boottime. Since SuSe has no rc.local like in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory (I assume it is like that if i want automated asterisk startup). Do you have any experience how this is implemented in SuSe, and if you have some useful script for starting asterisk, I would be very, i mean VERY pleased? Thank you all in advance! This mail sent through L-secure: http://www.l-secure.net/ ___ 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] XORCOM RAPID Asterisk - Suggestions?
We like it alot. It makes rapid deployment of asterisk boxes a breeze. Brent On 8/17/05, Sharadindu Mohanty [EMAIL PROTECTED] wrote: Hey Guys, Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now its fine as currently it is in testing stage with 3-4 users. Any Ideas??? Thanks Sharadindu Mohanty How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos ___ 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] realtime caching
We have a web interface where users can update their dialplan online (not in production yet). The web page modifies the mySQL record. It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -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] is this possible with asterisk?
Hello Everyone! I'm wondering if the following is possible with asterisk... What i'm trying to do is find a program or a solution that can help me set appointments for a delivery company... the program should call a person asking them if the following time is suitable for a delivery... if they agree, they press one and the system logs it... if they don't agree they press two, etc... Also, another thing the system would do, would be to call the person and ask them a couple of questions and have them rate the service by pressing 1, 2, 3, etc... If anybody can point me in the right direction, it would be highly appreciated... Thanks! ___ 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] snom hint
Hi list, anybody any example how to use it? I did not find any hint in the wiki nor in the mailinglist archive :-(. I want to use one button showing my agents the actual state (logged in or logged off) Thank you Gerd ___ 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] OT: PC network down if plugged in Polycom IP600
Hi all, I dont know why, but if I plug my PC inside the 'PC' slot on my polycom, this is not working. (Polycom IP600 is online on the net.) I'm using normal network cables. (I see jumpers behind the phone... do I need to play arround with that?) Any help would be appreciated. -- Alexandre Leclerc ___ 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] is this possible with asterisk?
Yes, you could do that with Asterisk and Cepstral/Festival. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 11, 2005 6:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] is this possible with asterisk? Hello Everyone! I'm wondering if the following is possible with asterisk... What i'm trying to do is find a program or a solution that can help me set appointments for a delivery company... the program should call a person asking them if the following time is suitable for a delivery... if they agree, they press one and the system logs it... if they don't agree they press two, etc... Also, another thing the system would do, would be to call the person and ask them a couple of questions and have them rate the service by pressing 1, 2, 3, etc... If anybody can point me in the right direction, it would be highly appreciated... Thanks! ___ 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] is this possible with asterisk?
Yes, its possible and not too difficult. You can start here to see what you can do with Call Files: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out And a simple example of this in action is the perl wake up call application: http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+wake-up Good luck! Tim [EMAIL PROTECTED] wrote: Hello Everyone! I'm wondering if the following is possible with asterisk... What i'm trying to do is find a program or a solution that can help me set appointments for a delivery company... the program should call a person asking them if the following time is suitable for a delivery... if they agree, they press one and the system logs it... if they don't agree they press two, etc... Also, another thing the system would do, would be to call the person and ask them a couple of questions and have them rate the service by pressing 1, 2, 3, etc... If anybody can point me in the right direction, it would be highly appreciated... Thanks! ___ 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] Any one using the new Digium echo cancellation cards
THe wiki doesn't seem to have any user reports. If your using them, how are the working, better, worse about the same. Also what hardware seems to be stable with them installed. Alan ___ 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] TxFax - RxFax on same machine hangs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 17 Aug 2005, Steve Underwood wrote: Bartek Kania wrote: If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. I can confirm that this problem appears on a call through the PSTN. My setup is: TxFax - Asterisk - E1 - Asterisk (same box) -RxFax Asterisk version 1.0.9 and spandsp version 0.0.2pre18 on debian woody (3.0). I sent you an email about it with some debug information a week or so ago. If you need it again, or need some other info I'll be happy to provide it. Did you put txfax in caller mode? Yes I did. This is a snippet from 'show channel' for the two channels: Name: Zap/3-1 Type: Zap ... Frames in: 5249 Frames out: 265 Time to Hangup: 0 Elapsed Time: 0h1m45s ... Application: RxFAX Data: /tmp/1123753288.12.tif Stack: 1 Blocking in: ast_waitfor_nandfds and Name: Zap/28-1 Type: Zap ... Frames in: 3123 Frames out: 430 Time to Hangup: 0 Elapsed Time: 0h1m3s ... Application: TxFAX Data: /usr/local/asterisk/var/spool/asterisk/faxspool//ff-psbj1x.tif|caller|debug Stack: 0 Blocking in: ast_waitfor_nandfds The console seems to indicate that the faxes start to communicate using the slow modems, and then hang after switching to a fast modem. Log is attached. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDA0I5ckvkFeO3ANARAjLJAJ0eXELd2WjzGOy59ptkFEE3kiUJcQCgxF9P 3WgYpTG5b1BfA3yOVk3w9wc= =lNab -END PGP SIGNATURE-Slow carrier up Slow carrier down Slow carrier up CSI: 40 35 38 33 20 30 30 30 36 2d 30 34 2d 36 34 2b 20 20 20 20 20 CSI without final frame tag Remote fax gave CSI as: +xx-xx- xxx DIS: 80 00 ce f4 80 80 81 80 80 80 18 DIS with final frame tag In state 10 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 c6 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 DCS with final frame tag In state 9 Coarse carrier frequency 1699.85 (66) Training error 0.506731 Training succeeded (constellation mismatch 0.703194) Changed from phase 6 to 3 Slow carrier up T4 timeout in state 4 Start rx document Start rx page - compression 2 ___ 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: PC network down if plugged in Polycom IP600
Alexandre Leclerc wrote: Hi all, I dont know why, but if I plug my PC inside the 'PC' slot on my polycom, this is not working. (Polycom IP600 is online on the net.) I'm using normal network cables. (I see jumpers behind the phone... do I need to play arround with that?) Any help would be appreciated. Try a different phone. I have one Polycom IP300 that will take down the whole network switch when attached for a few days. I have to return it to Polycom. -- 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] gnugk and asterisk
Hello there. Does anyone have idea how to setup these two to work together? I'm really going insane with this combination... Any .conf files or something? Cheers, Vedran. ___ 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] SIP message 183 and in band info
Hello, I have such a problem. I have an * configured as a peer connected to the gateway to PSTN. While calling to the switched off cell phone, the gateway sends to the * the SIP message 180 with the SDP part, and also a lot of rtp packets containing the operator's in band info. But * forwards the 180 to the UAC without the sdp part and also without the rtp stream. Is there any way, how to setup the * dialplan to translate all incoming 180 SIP messages to 183 with the SDP part and also to forward the rtp stream to the UAC?? Thanks for advices... Tomas ___ 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] is this possible with asterisk?
There is a .php wakeup agi on voip-info too. I don't think it would be that difficult to modify it to your needs On Wed, 2005-08-17 at 06:54, Tim Pushor wrote: Yes, its possible and not too difficult. You can start here to see what you can do with Call Files: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out And a simple example of this in action is the perl wake up call application: http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+wake-up Good luck! Tim [EMAIL PROTECTED] wrote: Hello Everyone! I'm wondering if the following is possible with asterisk... What i'm trying to do is find a program or a solution that can help me set appointments for a delivery company... the program should call a person asking them if the following time is suitable for a delivery... if they agree, they press one and the system logs it... if they don't agree they press two, etc... Also, another thing the system would do, would be to call the person and ask them a couple of questions and have them rate the service by pressing 1, 2, 3, etc... If anybody can point me in the right direction, it would be highly appreciated... Thanks! ___ 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 -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- 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] Voicemail Retrival
you could declare the phone names as variables..
PHONE1=SIP/phone1
PHONE1VM=12345
On Wed, 2005-08-17 at 03:31, Rudolf Ladyzhenskii wrote:
Hi,
This procedure will work under one condition -- your user names are
same as your extension numbers. I have same problem. I was giving
phones alphanumeric user names, like phone1.
When VoicemailMain is called with ${CALLERIDNUM}, it is actually
called as VoiceMailMain(phone1). As a result, voice mail is asking
for a mailbox number which is same as your extension number. (BTW, is
there a way to extract extension number rather than phone name?).
As I am experimenting with *, I will rename phones to match their
extensions.
Rudolf
- Original Message -
From: Sharadindu Mohanty
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, August 17, 2005 8:32 PM
Subject: Re: [Asterisk-Users] Voicemail Retrival
I did the same way but it is asking for some password and
mailbox. I think mail box is extension no but what abt
password?
Can i overide this procedure?
Thanks
Christoph Eicke [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 10:29, Sharadindu Mohanty
wrote:
Hi,
Hi!
Any ideas??
Yes, I do it in the following way. In extension.conf
add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then
automatically gets directed to their
voicemail where they have some options.
I hope this helps,
Christoph
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PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y
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RE: [Asterisk-Users] gnugk and asterisk
Hey Vedran I did this a while ago but to put you on the write track you have to register your gatekeeper (gnugk) with Asterisk as a gateway specifying a prefix, let's say for arguments sake '0'. Then any numbers dialled on your GK-managed H.323 network, that start with a zero, are routed to the gateway (in this case asterisk) If you still have problems I may be able to dig up some configs for you?? Cheers Jason Jason Penton PhD Candidate Department of computer Science Rhodes University Tel: +27 46 603 8640 Mobile: +27 82 376 6811 VoIP: sip:[EMAIL PROTECTED] Email: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vedran Dakic Sent: 17 August 2005 04:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] gnugk and asterisk Hello there. Does anyone have idea how to setup these two to work together? I'm really going insane with this combination... Any .conf files or something? Cheers, Vedran. ___ 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] gnugk and asterisk
Man, I would really be grateful if you could put me out of my misery and send me something, I don't know where's anything anymore in the config files or anything. Too much editing those in the past 16 hours, I guess.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Penton Sent: Wednesday, August 17, 2005 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] gnugk and asterisk Hey Vedran I did this a while ago but to put you on the write track you have to register your gatekeeper (gnugk) with Asterisk as a gateway specifying a prefix, let's say for arguments sake '0'. Then any numbers dialled on your GK-managed H.323 network, that start with a zero, are routed to the gateway (in this case asterisk) If you still have problems I may be able to dig up some configs for 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] Voicemail Retrival
There is a different approach to this;
Put a priority 'a' in the extension dialplan that goes to
Voicemmailmain(${EXTEN})
Users then dial there own extension from any location and press the *
key once voicemail picks up.
This method seems to emulate what most people are already used to.
If you have a voicemail button on the phone the other method works as
well, you can use both.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Wednesday, August 17, 2005 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail Retrival
you could declare the phone names as variables..
PHONE1=SIP/phone1
PHONE1VM=12345
On Wed, 2005-08-17 at 03:31, Rudolf Ladyzhenskii wrote:
Hi,
This procedure will work under one condition -- your user names are
same as your extension numbers. I have same problem. I was giving
phones alphanumeric user names, like phone1.
When VoicemailMain is called with ${CALLERIDNUM}, it is actually
called as VoiceMailMain(phone1). As a result, voice mail is asking
for a mailbox number which is same as your extension number. (BTW,
is
there a way to extract extension number rather than phone name?).
As I am experimenting with *, I will rename phones to match their
extensions.
Rudolf
- Original Message -
From: Sharadindu Mohanty
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, August 17, 2005 8:32 PM
Subject: Re: [Asterisk-Users] Voicemail Retrival
I did the same way but it is asking for some password and
mailbox. I think mail box is extension no but what abt
password?
Can i overide this procedure?
Thanks
Christoph Eicke [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 10:29, Sharadindu
Mohanty
wrote:
Hi,
Hi!
Any ideas??
Yes, I do it in the following way. In extension.conf
add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then
automatically gets directed to their
voicemail where they have some options.
I hope this helps,
Christoph
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PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y
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[Asterisk-Users] OH323 call leg and IAX call leg
Hi, I am having a strange problem. When ever I made a call, one leg is IAX and other leg is OH323. The call establish fine but anybody talking from OH323 leg side, I hear broken sound in IAX side. Something is wrong with RTP. Is it something do with FRAME set in OH323? if so, what will be the correct set? I am using Codec 729 for both call legs. Anybody has any idea on this issue? Thanks CM Rahman Jr. CCS Internet www.ccsi.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] realtime caching
It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -Matthew What would it take (you, $) to add functionality that is a cross between caching and not, that is it caches with a flag in the extension, so if the flag is present realtime will be queried even though the extension is in cache when a new call comes IN TO that extension. Outgoing calls would not really need a re-query unless something about the provisioning of the phone changes, at which point it would re-register anyways, right? The goal is caching for MWI and NAT but realtime for calling, so the database is checked on every inbound call in case the dialplan changed, and the cache updated accordingly. Maybe a TTL flag, and when the TTL expires the cache entry stays, but is re-queried when a dialplan match is found. The admin could then tune the performance by setting different TTLs, maybe 15 minutes for lightly loaded systems, 4 hours for heavy loaded systems. Dynamic updates take place in whatever timeframe is specified on the TTL or less. Have I missed something, is this functionality already present? Damon ___ 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] realtime caching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Wednesday, August 17, 2005 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] realtime caching It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -Matthew What would it take (you, $) to add functionality that is a cross between caching and not, that is it caches with a flag in the extension, so if the flag is present realtime will be queried even though the extension is in cache when a new call comes IN TO that extension. Outgoing calls would not really need a re-query unless something about the provisioning of the phone changes, at which point it would re-register anyways, right? The goal is caching for MWI and NAT but realtime for calling, so the database is checked on every inbound call in case the dialplan changed, and the cache updated accordingly. Maybe a TTL flag, and when the TTL expires the cache entry stays, but is re-queried when a dialplan match is found. The admin could then tune the performance by setting different TTLs, maybe 15 minutes for lightly loaded systems, 4 hours for heavy loaded systems. Dynamic updates take place in whatever timeframe is specified on the TTL or less. Have I missed something, is this functionality already present? Damon ___ I may have answered my own question, is it true that realtime extensions are still queried every call, and only chan_sip is effected by rtcachefriends? Damon ___ 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] TxFax - RxFax on same machine hangs
Hi Bartek, I posted the exact same problem last week - I found that if I connected two Asterisk systems together via a PRI crossover cable and talk txfax to rxfax then you get a T4 state timeout. I tried connecting ports one and two together on a TE410p and also connecting a TE410p to a TE110p, and a TE110p to a TE110p on different machines. I also found this when looping back via the PSTN. I read up a bit on what T4 actually is, and it seems to be a pretty high level state, where the faxes are transferring or about to transfer the tiff image data between themselves. The faxes will eventually hangup on each other - if you do a zap destroy channel to force hangup then you will sometimes get a segfault and asterisk will crash. I haven't found a solution for it, but it's not a big problem for me as I was only going txfax to rxfax as part of testing something else and I am using a hardfax attached to a SIP ata instead that works just fine against both rxfax and txfax. Craig - Original Message - From: Bartek Kania [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 17, 2005 9:57 PM Subject: Re: [Asterisk-Users] TxFax - RxFax on same machine hangs -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 17 Aug 2005, Steve Underwood wrote: Bartek Kania wrote: If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. I can confirm that this problem appears on a call through the PSTN. My setup is: TxFax - Asterisk - E1 - Asterisk (same box) -RxFax Asterisk version 1.0.9 and spandsp version 0.0.2pre18 on debian woody (3.0). I sent you an email about it with some debug information a week or so ago. If you need it again, or need some other info I'll be happy to provide it. Did you put txfax in caller mode? Yes I did. This is a snippet from 'show channel' for the two channels: Name: Zap/3-1 Type: Zap ... Frames in: 5249 Frames out: 265 Time to Hangup: 0 Elapsed Time: 0h1m45s ... Application: RxFAX Data: /tmp/1123753288.12.tif Stack: 1 Blocking in: ast_waitfor_nandfds and Name: Zap/28-1 Type: Zap ... Frames in: 3123 Frames out: 430 Time to Hangup: 0 Elapsed Time: 0h1m3s ... Application: TxFAX Data: /usr/local/asterisk/var/spool/asterisk/faxspool//ff-psbj1x.tif|caller|debug Stack: 0 Blocking in: ast_waitfor_nandfds The console seems to indicate that the faxes start to communicate using the slow modems, and then hang after switching to a fast modem. Log is attached. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDA0I5ckvkFeO3ANARAjLJAJ0eXELd2WjzGOy59ptkFEE3kiUJcQCgxF9P 3WgYpTG5b1BfA3yOVk3w9wc= =lNab -END PGP 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 ___ 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] Automatic start with SuSe linux
On Wednesday 17 August 2005 7:27 am, [EMAIL PROTECTED] wrote: Hi! I'm trying to start asterisk at boottime. Since SuSe has no rc.local like in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory (I assume it is like that if i want automated asterisk startup). Do you have any experience how this is implemented in SuSe, and if you have some useful script for starting asterisk, I would be very, i mean VERY pleased? To make it start on boot: insserv asterisk Start it immediately with: rcasterisk start -- James Oakley Engineering - SolutionInc Ltd. [EMAIL PROTECTED] http://www.solutioninc.com ++ This e-mail is CONFIDENTIAL and contains information intended only for the person(s) named. Any other distribution, copying or disclosure is strictly prohibited. If you have received this e-mail in error, please notify me immediately at 902 420 0077 or reply by e-mail to the sender and destroy the original communication. 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] PLEASE REPLY, are you using an X101P
If you don't mind sharing, what was the vendor that worked great? Thanks! On 8/17/05, VoIP Newbie [EMAIL PROTECTED] wrote: I bought 3 from 3 different vendors. One of them has echo issue. Another one has an issue regarding PCI master abort. Only one really works fine for me. These 3 cards use AMBIENT chip but with different layouts and SLICs. On 8/4/05, Mark Burton [EMAIL PROTECTED] wrote: X101P with Ambient md3200 chip on it, with the zaptel wcfxo driver Just an indication of how many people have got this to work would be useful. 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 ___ 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] realtime caching
pruning breaks asterisk on high loads at least on all 5 of our servers. all using different versions and custom. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. On 8/17/05, Damon Estep [EMAIL PROTECTED] wrote: It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -Matthew What would it take (you, $) to add functionality that is a cross between caching and not, that is it caches with a flag in the extension, so if the flag is present realtime will be queried even though the extension is in cache when a new call comes IN TO that extension. Outgoing calls would not really need a re-query unless something about the provisioning of the phone changes, at which point it would re-register anyways, right? The goal is caching for MWI and NAT but realtime for calling, so the database is checked on every inbound call in case the dialplan changed, and the cache updated accordingly. Maybe a TTL flag, and when the TTL expires the cache entry stays, but is re-queried when a dialplan match is found. The admin could then tune the performance by setting different TTLs, maybe 15 minutes for lightly loaded systems, 4 hours for heavy loaded systems. Dynamic updates take place in whatever timeframe is specified on the TTL or less. Have I missed something, is this functionality already present? Damon ___ 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] Voicemail crashes asterisk
When a user dial voicemail and just hangs up or enters the wrong password 3 times asterisk will crash. We are using Cisco 7960G with SIP My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC Any help would be great!!! Thanks ___ 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: PC network down if plugged in Polycom IP600
Chris Mason (Lists) a écrit : Alexandre Leclerc wrote: Hi all, I dont know why, but if I plug my PC inside the 'PC' slot on my polycom, this is not working. (Polycom IP600 is online on the net.) I'm using normal network cables. (I see jumpers behind the phone... do I need to play arround with that?) Any help would be appreciated. Try a different phone. I have one Polycom IP300 that will take down the whole network switch when attached for a few days. I have to return it to Polycom. I only bought one to test before buying some more... :) But thanks for this hint. I'll contact Polycom. Regards. -- Alexandre Leclerc ___ 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] snom hint
It's in the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom About halfway down the page where it says: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/17/05, Gerd Mueller [EMAIL PROTECTED] wrote: Hi list, anybody any example how to use it? I did not find any hint in the wiki nor in the mailinglist archive :-(. I want to use one button showing my agents the actual state (logged in or logged off) Thank you Gerd ___ 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 -- Tom ___ 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] Voicemail crashes asterisk
It was fixed a while ago, download new code. There is a bug in the tracker on it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, August 17, 2005 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicemail crashes asterisk When a user dial voicemail and just hangs up or enters the wrong password 3 times asterisk will crash. We are using Cisco 7960G with SIP My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC Any help would be great!!! Thanks ___ 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] problems with eyebeam - video phone
quickly this looks like a incompatible codec.. or unrecognized.. show codecs on CLI show show 262144 (1 18) (0x4) videoh261 (H.261 Video) 524288 (1 19) (0x8) videoh263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) does it ? On 8/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thank you for your answer. I didn't register on the domain of the Eyebeam software, actually I don't understand how to do that! I bouught 5 eyebeam activation keys and I am trying with the first 2 of them On the Eyebeam side (both eyebeam), I only enabled the Basic H.263 codec, no other. If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the two video phone speak without any problem (but without any video) If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the first video phone call the second, the second answer and immediately the call ends. If Ilook at /var/log/asterisk/full, I see: Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi completed, returning 0 Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial(SIP/551-eac0, SIP/552|25|tr) in new stack Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL) Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0 Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0 Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552 Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0 Aug 17 08:37:06 VERBOSE[14731]: -- Called 552 Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102 Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0 Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2) to SIP/552-ff46(524288) Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make SIP/551-eac0 compatible with SIP/552-ff46 Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse counter It seems the problem documented in bug http://bugs.digium.com/bug_view_page.php?bug_id=0003709 but actually it is not exactly the same. moreover: is there any way to put the patch described in http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *) in asterisk 1.0.9 and not asterisk CVS HEAD ? Any help will be greatly appreciated. Andrea Carlos Alperin [EMAIL PROTECTED] om.netTo Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 16/08/2005 20.48 RE: [Asterisk-Users] problems with eyebeam - video phone Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi, I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I only use H.263 and SIP. (G.729) Now, the more important question is if you register on the domain on the Eyebeam software. I found that this was the full secret about this. Let me know your configuration on the Eyebeam side. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 11:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problems with eyebeam - video phone I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ; allow=h263p ; H.263p is the enhanced video codec context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf And I left only H.263 basic in codec's configuration in Video Phone. No chance to get the communication
Re: [Asterisk-Users] problems with eyebeam - video phone
Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats is from what you pasted btw.. Don't know any of 0x8 formats is 524288 (1 19) (0x8) videoh263 (H.263 Video) meaning it downst understand it or find it On 8/17/05, Jimmy Smith [EMAIL PROTECTED] wrote: quickly this looks like a incompatible codec.. or unrecognized.. show codecs on CLI show show 262144 (1 18) (0x4) videoh261 (H.261 Video) 524288 (1 19) (0x8) videoh263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) does it ? On 8/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thank you for your answer. I didn't register on the domain of the Eyebeam software, actually I don't understand how to do that! I bouught 5 eyebeam activation keys and I am trying with the first 2 of them On the Eyebeam side (both eyebeam), I only enabled the Basic H.263 codec, no other. If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the two video phone speak without any problem (but without any video) If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the first video phone call the second, the second answer and immediately the call ends. If Ilook at /var/log/asterisk/full, I see: Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi completed, returning 0 Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial(SIP/551-eac0, SIP/552|25|tr) in new stack Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL) Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0 Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0 Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552 Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0 Aug 17 08:37:06 VERBOSE[14731]: -- Called 552 Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102 Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0 Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2) to SIP/552-ff46(524288) Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make SIP/551-eac0 compatible with SIP/552-ff46 Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse counter It seems the problem documented in bug http://bugs.digium.com/bug_view_page.php?bug_id=0003709 but actually it is not exactly the same. moreover: is there any way to put the patch described in http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *) in asterisk 1.0.9 and not asterisk CVS HEAD ? Any help will be greatly appreciated. Andrea Carlos Alperin [EMAIL PROTECTED] om.netTo Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 16/08/2005 20.48 RE: [Asterisk-Users] problems with eyebeam - video phone Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi, I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I only use H.263 and SIP. (G.729) Now, the more important question is if you register on the domain on the Eyebeam software. I found that this was the full secret about this. Let me know your configuration on the Eyebeam side. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 11:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problems with eyebeam - video phone I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=h263
[Asterisk-Users] Re: Automatic start with SuSe linux
On Wed, Aug 17, 2005 at 02:11:09PM +0100, Angus Comber wrote: You could just add the line asterisk to /etc/init.d/boot.local Excerpt from /etc/init.d/boot.local # Here you should add things, that should happen directly after # booting # before we're going to the first run level. Do not attempt to start asterisk here. There is some SuSE asterisk rpm available for SuSE 9.3. It is asterisk 1.0.6, but you can extract the boot script without installing anything else. rpm2cpio asterisk-1.0.6-4.i586.rpm | cpio -i -d -v './etc/init.d/asterisk' Modify the script and copy it to /etc/init.d/. SuSE program insserv can be used to add the symlinks to /etc/init.d/rc3.d/ directory. -- Stefan Tichy [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] DID on TDM400P Question?
Does anyone know if the current TDM400 card can take DID digits from the LEC? If so is there any reference to how to set this all up? As I get my current service from my LEC over an IAD, so would be sweet to just have trunks, not each channel specific to a number. Also if the above is possible, if the line is being used for DID, then is this only workable for inbound, or can I also seize the line and use it for outbound calls. I know with PRI's that is easy, but never had to play with this on an analog port level. Just having a PRI at home isn't practical, so not something I can really do. Any input, or ideas on this would be most appreciated.. Thanks... --- Howard Leadmon http://www.leadmon.net ___ 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] realtime caching
Damon Estep wrote: I may have answered my own question, is it true that realtime extensions are still queried every call, and only chan_sip is effected by rtcachefriends? Damon True. RealTime Exensions are queried every time. There is no caching of extensions. If you turn on debug log, you can watch each query. -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] realtime caching
Jimmy Smith wrote: pruning breaks asterisk on high loads at least on all 5 of our servers. all using different versions and custom. You should bug report this if you have a backtrace. Kevin and I worked on the pruning stuff (well, he coded and i tested) for a while and seemedly got it working. -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] SIP agent phone w/ headset
Thanks for the feedback Just for a background, one of the reasons for redundancy (notice the quotes ;-) is that the PC is setup as a kiosk style application in which we do a shell replacement with the Windows Explorer, so instead of a desktop, the user gets a dedicated application which is very thin client like. The reason we're wary of integrating voice in this application, which is certainly doable with one of the various SDK's out there, is that we also host an Oracle forms client interface as part of this and this thing is a big ol' pig and screws up the PC on a regular basis (it's not my product so there's not much I can do but cope with it). Anyway, the users are very low level users and do not know much about PC's, so at the slightest hint of an issue they just punch the reset button on the pc and reboot it (or unplug it, or...you get the idea). Yes, your assumptions are correct in that these agents are in a receive only situation, with very limited call function capabilities. The end goal is that the software client running on the PC will be able to control the extension and act like a manager for that phone unit. I'm probably asking to stretch what is out in the market right now, but I'd be remiss for not looking. Most likely, we'll end up with a soft phone embedded in the client software, but I'm not looking forward to dealing with USB headsets. Colin Stefani Tideworks Technology -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Tuesday, August 16, 2005 4:23 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP agent phone w/ headset On 16:01, Tue 16 Aug 05, Colin Stefani wrote: I have a call center where we're looking at converting it from a traditional PBX w/ digital phone agent sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad, buttons or handset, just a line input and a headset jack. I need the equivalent. I know the first thing you think is why don't you use the agent's PC as the VoIP client and do a softphone, however I need to protect the caller from getting cut off should the PC crash/die/etc. While paranoid it's something where a regular endpoint like an ATA or SIP phone would be the best option. SIP phones and ATA's can die too. * can die too heck even your power can go down (hurricane, terrorist attack, etc, etc) A properly configured pc with a softfone can be as stable as a normal phone, it all depends what the users are doing with it (I have had bad experience with pc's where users can install their own stuff etc). I have a workstation with an uptime of over 500 days. This email was written on it. The problem will be the 'without keyped, buttons or handset'. I'm not aware of a SIP device that has only a line button and a headset and nothing else. Judging on the setup you outlined, the agents are not able to transfer the call to admin/other_user/parking_slot. They are only able to receive calls, and that's all. If so, you can create them as 'user' only in sip.conf That way they are only able to receive calls, but not make calls. The interface to * is something you choose. Of course phones/ATA's are less error-sensitive as pc's, cause you can configure them. Just make sure noone can guess the username/password for the ATA/phone config interface. Hope this helps, -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Avaya 4602 SIP Internal Dial Plan
Hi, I'm trying to disable the internal dial plan of an Avaya 4602 with SIP firmware 1.1 but couldn't find how to do it. Even if I configure a custom Dial Plan it keeps adding other builtin rules to my dial plan. Ex: Configured dial plan: DialPlan19xx|7[8-9]xx|0xxx+ Reboot. On the syslog it shows: Aug 17 12:42:12 192.168.0.115 DigitMap: 19xx|7[8-9]xx|0xxx+|xx+*|xx+#|*0[1-9]|*1[0-9]|*2[0-5]|*6[0189]|*7[0-35]|*74 The phone mix my dial plan with this rules: xx+*|xx+#|*0[1-9]|*1[0-9]|*2[0-5]|*6[0189]|*7[0-35]|*74 Anyone knows how to disable this ? No info on Administrator's Guide about this hardcoded dial plan :) Thanks, Leonardo ___ 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] New Astmanproxy Mailing List, and New Version 1.11
Greetings -- Many of you have downloaded and tried out Astmanproxy, a multi-threaded C-based proxy for Asterisk's Manager Interface. It has been under development since April 2005 and was presented at the Madrid Astricon in June, and will also be presented at Astricon in Anaheim in October. There has been interest in setting up an Astmanproxy mailing list specifically devoted to discussion of astmanproxy and general Asterisk API topics. This list is now ready to use! Astmanproxy Mailing List Subscribe: [EMAIL PROTECTED] Post message: [EMAIL PROTECTED] List owner: [EMAIL PROTECTED] Unsubscribe:[EMAIL PROTECTED] Additionally, a new version of Astmanproxy (1.11) is now available from www.popvox.com/astmanproxy . A couple of small bugs are fixed, and a few new features have been added. Astmanproxy can communicate with multiple Asterisk servers, and can act as a single point of contact for your applications to communicate with Asterisk. Multiple input/output formats are supported, including Standard, XML, HTTP, and CSV. All I/O handlers are implemented via an extensible interface to allow for easy support of new formats. Long term, astmanproxy is intended to serve as a central piece of glue between Asterisk servers and applications, and is a good structure in which to implement new APIs. Astmanproxy takes load off of Asterisk by communicating with your applications and leaving Asterisk free to do what it does best, telephony. Please join in the Astmanproxy discussion and let us know what features you'd like to see added! Regards, David Troy -- David C. Troy President/CEO popvox, LLC [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] Any success with Polycom DHCP VLAN discovery?
Greetings. Has anyone made this work with BootROM 2.6.2 and app 1.5.2? I've tried sending DHCP options 128, 144, 157 and 191 containing a single digit (the VLAN ID) with the phone's 'Fixed' setting for DHCP VLAN discovery. Different DHCP data types don't seem to help, as I've tested with raw bytes, ASCII and 16-bit unsigned ints to no avail. Setting the phone's option to 'Custom' and using option 129 hasn't worked, nor has enabling the CDP option (totally different, I realize). Any suggestions? Thanks, Tim ___ 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] canreinvite in sip.conf
Hi, Im using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two phones) Id like to known if it can works. Does anyone just did it? Thanks in advance Gio ___ 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] Iaxy Distinctive Ring
Is there a way to cause an Iaxy to do distinctive ring? Clint ___ 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] Can not dial more then 23 calls
It looks like you are sending calls out over one port. To help you out, we
will need to look at your extensions.conf and zapata.conf. My hunch is that
you are dialing out using something like
Dial(zap/g3/${EXTEN},20,) where the group of channels you're using is on one
port of your Digium card.
If my math is right, you should be able to send 69 calls long distance, and 23
local calls at a time with no failover.
Louie Tarpo
Adam Aircraft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Pudenz,
Duane
Sent: Wednesday, August 17, 2005 12:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can not dial more then 23 calls
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test number out the PRI line.
We can not get more then 23 total active calls to connect to the test
numbers, the test numbers terminate to another PBX that we can monitor.
We have dialed out using cell phones to this other PBX while the test is
happening and it connects, meaning it has more then 23 active calls on
it.
If we place more then 23 calls then it seems to 'queue' the extra calls,
though not all of the extra calls complete after we stop adding new
calls. They seem to get stuck in a queue or lost. We will send 200
calls through the Asterisk server and all but about 20 do eventually
complete. Those 20 or so are stuck as Asterisk thinks the channels are
busy with the calls when in fact there are no 'real' calls on the
server.
We can send 30 calls through the LD or PRI and only 23 are actually
connected at a time. We can send 30 calls to both LD and PRI at the
same time and still only a mixture of 23 calls are actually active at
one time.
So our issue seems to be located in our Asterisk server. Is there a way
to limit or throttle an Asterisk server so that it will not place more
then 'x' calls?
We need to be able to support 48 calls.
Any ideas?
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Re: [Asterisk-Users] Iaxy Distinctive Ring
I will answer you, the same somebody told me at IIRC. A watch has more processor power than a Iaxy... So, in few words: No. I already tried to have a lot ot things (callpickup, distinctive ring, changing the time of flash pulse) and nothing... El Miércoles, 17 de Agosto de 2005 11:33, Clint Guillot escribió: Is there a way to cause an Iaxy to do distinctive ring? Clint ___ 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] Iaxy Distinctive Ring
2005/8/17, Andrès Tello Abrego [EMAIL PROTECTED]: I will answer you, the same somebody told me at IIRC. A watch has more processor power than a Iaxy... Uuuuh... well, I feel stupid but... what is the meaning of laxy ? 'cause... a watch... ;o))) sorry for my ignorance... Best regards, YLB. [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] SIP agent phone w/ headset
Colin, Is there any reason why you couldn't just set up a T1 card and channel banks (as many as needed) and use your exisiting agent phones via zap channels? Tom On Aug 17, 2005, at 11:59 AM, Colin Stefani wrote: Thanks for the feedback Just for a background, one of the reasons for redundancy (notice the quotes ;-) is that the PC is setup as a kiosk style application in which we do a shell replacement with the Windows Explorer, so instead of a desktop, the user gets a dedicated application which is very thin client like. The reason we're wary of integrating voice in this application, which is certainly doable with one of the various SDK's out there, is that we also host an Oracle forms client interface as part of this and this thing is a big ol' pig and screws up the PC on a regular basis (it's not my product so there's not much I can do but cope with it). Anyway, the users are very low level users and do not know much about PC's, so at the slightest hint of an issue they just punch the reset button on the pc and reboot it (or unplug it, or...you get the idea). Yes, your assumptions are correct in that these agents are in a receive only situation, with very limited call function capabilities. The end goal is that the software client running on the PC will be able to control the extension and act like a manager for that phone unit. I'm probably asking to stretch what is out in the market right now, but I'd be remiss for not looking. Most likely, we'll end up with a soft phone embedded in the client software, but I'm not looking forward to dealing with USB headsets. Colin Stefani Tideworks Technology -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Tuesday, August 16, 2005 4:23 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP agent phone w/ headset On 16:01, Tue 16 Aug 05, Colin Stefani wrote: I have a call center where we're looking at converting it from a traditional PBX w/ digital phone agent sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad, buttons or handset, just a line input and a headset jack. I need the equivalent. I know the first thing you think is why don't you use the agent's PC as the VoIP client and do a softphone, however I need to protect the caller from getting cut off should the PC crash/die/etc. While paranoid it's something where a regular endpoint like an ATA or SIP phone would be the best option. SIP phones and ATA's can die too. * can die too heck even your power can go down (hurricane, terrorist attack, etc, etc) A properly configured pc with a softfone can be as stable as a normal phone, it all depends what the users are doing with it (I have had bad experience with pc's where users can install their own stuff etc). I have a workstation with an uptime of over 500 days. This email was written on it. The problem will be the 'without keyped, buttons or handset'. I'm not aware of a SIP device that has only a line button and a headset and nothing else. Judging on the setup you outlined, the agents are not able to transfer the call to admin/other_user/parking_slot. They are only able to receive calls, and that's all. If so, you can create them as 'user' only in sip.conf That way they are only able to receive calls, but not make calls. The interface to * is something you choose. Of course phones/ATA's are less error-sensitive as pc's, cause you can configure them. Just make sure noone can guess the username/password for the ATA/phone config interface. Hope this helps, -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup? op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] (no subject)
Hi: I was running TDM12B. Both FXS and FXO were working fine. Then all of the sudden FXS had problems. When I pick-up the phone and dial any number, FXS doesn't respond. I just keep hearing the normal signaling line tone comming from the FXS. I changed the FXS module and it had the same problem. I changed the the TDM card and installed different FXS and nothing changed. I appreciate any suggestions. Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] Xten Digum TDP FXO card: No sound
I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm doing wrong? TIA ___ 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] Does intel 865 board works fine with Asterisk
Hi: I would like to know what are the issues I need to look for in a chipset board so I can make sure it works fine with digium cards and Asterisk . Is intel board 865 fits the description? Regards; __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] Iaxy Distinctive Ring
A iaxy, is a CPE device that provides VOIP capabilities to normal phones, using the iax protocol... So is a little hardware, for telephony usages, which doesn't have a lot of features, and is't so cheap... El Miércoles, 17 de Agosto de 2005 11:54, Yoann Le Bihan escribió: 2005/8/17, Andrès Tello Abrego [EMAIL PROTECTED]: I will answer you, the same somebody told me at IIRC. A watch has more processor power than a Iaxy... Uuuuh... well, I feel stupid but... what is the meaning of laxy ? 'cause... a watch... ;o))) sorry for my ignorance... Best regards, YLB. [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] FXO port trhoug optimum voice VOIP service
Sorry it took me so long to keep on this thread. But I got a quation Rich. Can the impedance missmatch kill the dial tone completely? This is, when I plug my X100p clone card to my line the dial tone just goes away. I check this by using an analog phone that is also on the line. Is it possible to fix this by using the rx/tx in the zaptel configuration? Maybe I need a different signalling since I'm actually behind an VoIP - analog adapter? Any help would be appreciated. Carlos --- Rich Adamson [EMAIL PROTECTED] wrote: Based on research that I did some time ago, there are multiple versions of the MD3200 chipset. One targeted for use in US telephone systems, and another targeted for non-US systems (that have different impedence matching requirements). Sounds like you have one of each. I have 2 OEM X100P. The one from www.broad-tel.com works fine.However, the other one has echo. Both use MD3200 chips. Any one knows why it is so?? On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote: Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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 Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ 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 ---End of Original Message- ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] Asterisk and Port
[EMAIL PROTECTED] ~]# netstat -naptu | grep asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 9231/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 9231/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 9231/asterisk udp0 0 xx.yy.zz.ww:50600.0.0.0:* 9231/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 9231/asterisk Hi, My asterisk server is listening to the above ports. would somebody explain, what ports are for what. Is there any security issue with these ports? what firewall messure you do regarding these open ports? Thanks___ 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] Voicemail file permissions
On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote: Is there a way around this w/o giving everyone root privileges! Do you want to allow every user to delete another user's voicemail? If not, how do you sync voicemail users and samba users? I want each user to see, read and write (delete) their own voicemail ONLY (i.e. a user shouldn't be able to listen to someone elses voicemails). I gave each user an account on the Asterisk box and limited their access to their mailbox folder only. So don't waste your time on saving the voicemail on Asterisk. Save it on a specific folder in an imap server on the user's home directory. If you use a decent mail client, getting notifications for new mails in that folders, deleting them, playing them, and whatever should be easy. On the Asterisk side you only need to keep voicemail config in sync. Maybe it would be easier to just forward every mailbox nnn to [EMAIL PROTECTED] and use an aliases file to do the real forwarding. That way you keep the emails away from Asterisk's config. The downside: no message-waiting indicator. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Does intel 865 board works fine with Asterisk
I have one Asterisk system working with a Junghanns BRI card and another working with a Digium TDM card with an Intel D865 motherboard. Angus - Original Message - From: jonny hashem [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, August 17, 2005 6:14 PM Subject: [Asterisk-Users] Does intel 865 board works fine with Asterisk Hi: I would like to know what are the issues I need to look for in a chipset board so I can make sure it works fine with digium cards and Asterisk . Is intel board 865 fits the description? Regards; __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] FXS on TDM12B suddenly stopped working Properly
Hi: I was running TDM12B. Both FXS and FXO were working fine. Then all of the sudden FXS had problems. When I pick-up the phone and dial any number, FXS doesn't respond. I just keep hearing the normal signaling line tone comming from the FXS. I changed the FXS module and it had the same problem. I changed the the TDM card and installed different FXS and nothing changed. I appreciate any suggestions. Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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
