[asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone, once again - last time to publish this.. i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the snoms doesnt understand this and thinks the caller is still on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --ASTERISK IS INVITING MY SNOM AT HOME-- Audio is at [ I P - A S T E R I S K ] port 11576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP --SNOM IS ANSWERING THE CALL-- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The same as above.. --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS-- --- SIP read from [ I P - A N G E R U F E N E R ]:5060 --- BYE sip:[ TEL. CALLER ]...@[ I P - A S T E R I S K ] SIP/2.0 Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport From: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;tag=e8yr1936gy To: [ MyName in the Snom ], [ MyName in the Snom ], ;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt ) I would be glad if someone can take a look... Kindly regards, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the snoms doesnt understand this and thinks the caller is still on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --ASTERISK IS INVITING MY SNOM AT HOME-- Audio is at [ I P - A S T E R I S K ] port 11576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP --SNOM IS ANSWERING THE CALL-- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The same as above.. --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS-- --- SIP read from [ I P - A N G E R U F E N E R ]:5060 --- BYE sip:[ TEL. CALLER ]...@[ I P - A S T E R I S K ] SIP/2.0 Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport From: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;tag=e8yr1936gy To: [ MyName in the Snom ], [ MyName in the Snom ], ;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt ) I would be glad if someone can take a look... Kindly regards, Stefan begin:vcard fn:Stefan Agethen n:Agethen;Stefan org;quoted-printable:B=C3=A4ckerei Agethen;Verkauf adr:;;Alstadener Str. 137;Oberhausen;NRW;46049;Deutschland email;internet:staget...@baeckereiagethen.de title;quoted-printable:B=C3=A4ckermeister, Kaufmann tel;work:0208-84804-40 tel;fax:0208-84804-24 tel;home:0208-84804-41 tel;cell:0208-84804-40 x-mozilla-html:TRUE url:www.agethen.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the snoms doesnt understand this and thinks the caller is still on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --ASTERISK IS INVITING MY SNOM AT HOME-- Audio is at [ I P - A S T E R I S K ] port 11576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP --SNOM IS ANSWERING THE CALL-- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The same as above.. --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS-- --- SIP read from [ I P - A N G E R U F E N E R ]:5060 --- BYE sip:[ TEL. CALLER ]...@[ I P - A S T E R I S K ] SIP/2.0 Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport From: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;tag=e8yr1936gy To: [ MyName in the Snom ], [ MyName in the Snom ], ;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt ) I would be glad if someone can take a look... Kindly regards, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WANTED : Zaptel Patch - Dtmfthreshold
I am using Zaptel for my Wildcard and got DTMF Tones in my conversations since a long time. To control this , i search for a possibility to control the detection of DTMF and to control talkoff, called dtmfthreshold. As seen in mISDN* i am in need of an Patch to control the dtmfthreshold in zaptel, it must be possible, because there is an value, staticly set to 1000, the value could be controlled between x and 4000, thats the things i think to know. There seems to be no other way than writing a patch to set this value in a config file. So - is there anyone who could write such a method - my programming background is not the best to do this. I would be very happy and - of course - there are many others who could use this option in the future. I have tried MANY Solutions, so please - lets dont talk about relaxdtmf and so on ;) --misdn.org *mISDN -- copy of the FAQs found in misdn.org : *Why are my dtmf tones not detected everytime?* We've added a configurable *dtmftreshold*, the default is 100, it can have values between 20 and 500. It needs to be changed either by giving the module parametern dtmftreshold_option to the mISDN_dsp modul, or simply by setting the value *dtmftreshold*=xxx in the /etc/misdn-init.conf --misdn.org Greets from Germany to the Asterisk Community, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. I use rfc2833 for dtmf, alaw as codec. Yes, a lowering could be a idea, but the problem is logged on any kind of channels in my system, like zap, misdn, sip and iax. That is my problem :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? Thanks, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. I use rfc2833 for dtmf, alaw as codec. Yes, a lowering could be a idea, but the problem is logged on any kind of channels in my system, like zap, misdn, sip and iax. That is my problem :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pressing * makes Asterisk destroy my call
I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN Cards, if i press in a call the * Asterisk, Asterisk destroys the call not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell me in the warnings-log that the bridging was not successfull ?! If have disabled the function to hangup in the features.conf, but the key is still available, can someone explain me whats going on there ? Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi, I have asked this question months ago - i have toggled down all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like A 0 or something that do something like # or *. In my case, the * let Asterisk hang up my call, i searched for help, but nobody knows what to do - so i disabled the hangup feature and so on, but the problem still exists :( I sets the hangup-function to : == Remapping feature Disconnect Call (disconnect) to sequence '*0' My System is a : Asus with an AMD Athlon XP 3000+ with 512MB of RAM, 1 Wildcard TDM40B, 2 HFC ISDN PCI Cards from Acer (128k Surf). Installed is : Debian 3.1 with unstable packages to get Kernel 2.6.15-1 (AMD Kernel) (in earlier days my ISDN Driver, mISDN only works with Kernel 2.6.12 or higher, Debian is 2.6.8, so...) The needed Packages for Asterisk are installed (My Installation Step-by-Step in german is here : http://www.ip-phone-forum.de/showpost.php?p=657963postcount=7) Zaptel 1.2.9 Asterisk 1.2.12.1 mISDN in 0.3.0 RC 23 I have changed mpeg123 against madplay. The Problem exists since a half year or more, i like to say it in another way : i have RECOGNIZED the problem since a half year, i have done many updates of all packages and a clean install to merge this prob, no luck, it still exists. The facts i know about it : During such a * DTMF Shooting the logfiles recognized this (see the channel types!) : -- NOTICES -- Nov 6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels mISDN/1-1 and Zap/1-1 Nov 6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels SIP/40-0815e778 and SIP/pbx1-08281bc8 Nov 6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels SIP/40-0826c530 and IAX2/pbx1-1 DTMF Tone Log : Nov 6 05:00:33 DTMF[18215] channel.c: SIP/50-0824f1e0 : A Nov 6 08:44:05 DTMF[21660] channel.c: Zap/1-1 : A Nov 6 09:43:00 DTMF[22520] channel.c: SIP/pbx1-08274fb8 : 0 Nov 6 09:53:26 DTMF[22637] channel.c: Zap/1-1 : * Nov 6 10:05:28 DTMF[22902] channel.c: SIP/pbx1-08281bc8 : * Nov 6 10:14:42 DTMF[23288] channel.c: mISDN/2-1 : 8 Nov 6 10:16:11 DTMF[23426] channel.c: SIP/pbx1-08274690 : * Nov 6 10:17:45 DTMF[23288] channel.c: Zap/1-1 : A Nov 6 10:32:54 DTMF[23545] channel.c: Zap/1-1 : D Nov 6 10:35:58 DTMF[23792] channel.c: SIP/pbx1-08273ef8 : * -- ASTERISK SIP DEBUG (one case) -- Nov 6 10:35:54 DEBUG[23792] channel.c: Got DTMF on channel (SIP/40-0825b3c8) Nov 6 10:35:54 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:54 DEBUG[23792] res_features.c: Feature interpret: chan=SIP/40-0825b3c8, peer=SIP/pbx1-08273ef8, sense=1, features=18 Nov 6 10:35:54 DEBUG[23792] res_features.c: Set time limit to 500 Nov 6 10:35:55 DEBUG[23792] channel.c: Nobody there, continuing... Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/40-0825b3c8, c1=SIP/pbx1-08273ef8, flags: No,Yes,No,No Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:58 DEBUG[23792] res_features.c: Timed out for feature! Nov 6 10:35:58 DEBUG[23792] channel.c: Hanging up channel 'SIP/pbx1-08273ef8' Nov 6 10:35:58 DEBUG[23792] chan_sip.c: Hangup call SIP/pbx1-08273ef8, SIP callid [EMAIL PROTECTED]) Nov 6 10:35:58 DEBUG[23792] chan_sip.c: update_call_counter(02088480499) - decrement call limit counter Nov 6 10:35:58 DEBUG[23792] app_dial.c: Exiting with DIALSTATUS=ANSWER. Nov 6 10:35:58 DEBUG[23792] pbx.c: Spawn extension (voip_wahl,_X.,6) exited non-zero on 'SIP/40-0825b3c8' As you can see in the DTMF Log - there are many Digits send, but they dont scare me, but the * are disconnecting my calls - thats a problem for me and my business.. I HOPE !!! you can help me, Best wishes, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi, I have asked this question months ago - i have toggled down all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like A 0 or something that do something like # or *. In my case, the * let Asterisk hang up my call, i searched for help, but nobody knows what to do - so i disabled the hangup feature and so on, but the problem still exists :( I sets the hangup-function to : == Remapping feature Disconnect Call (disconnect) to sequence '*0' My System is a : Asus with an AMD Athlon XP 3000+ with 512MB of RAM, 1 Wildcard TDM40B, 2 HFC ISDN PCI Cards from Acer (128k Surf). Installed is : Debian 3.1 with unstable packages to get Kernel 2.6.15-1 (AMD Kernel) (in earlier days my ISDN Driver, mISDN only works with Kernel 2.6.12 or higher, Debian is 2.6.8, so...) The needed Packages for Asterisk are installed (My Installation Step-by-Step in german is here : http://www.ip-phone-forum.de/showpost.php?p=657963postcount=7) Zaptel 1.2.9 Asterisk 1.2.12.1 mISDN in 0.3.0 RC 23 I have changed mpeg123 against madplay. The Problem exists since a half year or more, i like to say it in another way : i have RECOGNIZED the problem since a half year, i have done many updates of all packages and a clean install to merge this prob, no luck, it still exists. The facts i know about it : During such a * DTMF Shooting the logfiles recognized this (see the channel types!) : -- NOTICES -- Nov 6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels mISDN/1-1 and Zap/1-1 Nov 6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels SIP/40-0815e778 and SIP/pbx1-08281bc8 Nov 6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels SIP/40-0826c530 and IAX2/pbx1-1 DTMF Tone Log : Nov 6 05:00:33 DTMF[18215] channel.c: SIP/50-0824f1e0 : A Nov 6 08:44:05 DTMF[21660] channel.c: Zap/1-1 : A Nov 6 09:43:00 DTMF[22520] channel.c: SIP/pbx1-08274fb8 : 0 Nov 6 09:53:26 DTMF[22637] channel.c: Zap/1-1 : * Nov 6 10:05:28 DTMF[22902] channel.c: SIP/pbx1-08281bc8 : * Nov 6 10:14:42 DTMF[23288] channel.c: mISDN/2-1 : 8 Nov 6 10:16:11 DTMF[23426] channel.c: SIP/pbx1-08274690 : * Nov 6 10:17:45 DTMF[23288] channel.c: Zap/1-1 : A Nov 6 10:32:54 DTMF[23545] channel.c: Zap/1-1 : D Nov 6 10:35:58 DTMF[23792] channel.c: SIP/pbx1-08273ef8 : * -- ASTERISK SIP DEBUG (one case) -- Nov 6 10:35:54 DEBUG[23792] channel.c: Got DTMF on channel (SIP/40-0825b3c8) Nov 6 10:35:54 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:54 DEBUG[23792] res_features.c: Feature interpret: chan=SIP/40-0825b3c8, peer=SIP/pbx1-08273ef8, sense=1, features=18 Nov 6 10:35:54 DEBUG[23792] res_features.c: Set time limit to 500 Nov 6 10:35:55 DEBUG[23792] channel.c: Nobody there, continuing... Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/40-0825b3c8, c1=SIP/pbx1-08273ef8, flags: No,Yes,No,No Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:58 DEBUG[23792] res_features.c: Timed out for feature! Nov 6 10:35:58 DEBUG[23792] channel.c: Hanging up channel 'SIP/pbx1-08273ef8' Nov 6 10:35:58 DEBUG[23792] chan_sip.c: Hangup call SIP/pbx1-08273ef8, SIP callid [EMAIL PROTECTED]) Nov 6 10:35:58 DEBUG[23792] chan_sip.c: update_call_counter(02088480499) - decrement call limit counter Nov 6 10:35:58 DEBUG[23792] app_dial.c: Exiting with DIALSTATUS=ANSWER. Nov 6 10:35:58 DEBUG[23792] pbx.c: Spawn extension (voip_wahl,_X.,6) exited non-zero on 'SIP/40-0825b3c8' As you can see in the DTMF Log - there are many Digits send, but they dont scare me, but the * are disconnecting my calls - thats a problem for me and my business.. I HOPE !!! you can help me, Best wishes, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: DTMF Tones occuring randomly
Stefan Agethen wrote: / Hi, // // I have asked this question months ago - i have toggled down all DTMF // Recognizations in my Asterisk (no more features etc) // and found more people which recognized the same problem, but i cant find // any help for them and me. / The problem is called Talk Off (or maybe Talkoff). Search the archives for that. Yes, i know, talkoff means that at example a human voice is interpreted as a DTMF Signal, i have searched for that, the only possible thing i´ve found is the statement to turn on dtmfthreshold - but this only worked versions ago in zapata.conf, i think. I cant find anymore Information to dtmfthreshold. No more helpful results any more for my talkoff... :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Linksys PAP2: calling tone stops after 5
Message: 7 Date: Sun, 29 Oct 2006 22:00:22 +0100 From: Jose Limeres [EMAIL PROTECTED] Subject: [asterisk-users] Linksys PAP2: calling tone stops after 5 tones To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose Hi Jose, i got the same expect that the tone stops after 2 tones, everything work well but the tone just stops. I´ve read something about setting the Sticky 183 to No ... check this please. In my case STicky is off - but the problem still exists :( Greets to you, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: ECHO Cancellation in SIP Calls
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with echocancel=... and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP ! Are you hearing the echo, or is the far end party? I can hear the Echo, the end party never got this problem. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: ECHO Cancellation in SIP Calls
Message: 7 Date: Thu, 26 Oct 2006 22:56:58 -0400 From: Michael Araba [EMAIL PROTECTED] Subject: [asterisk-users] RE: ECHO Cancellation in SIP Calls To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I am surprised that you are getting echo on SIP calls. You can get echo in two scenarios on SIP calls. 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need to enable echo canceller and AGGRESSIVE if needed in zconfig.h. 2. Second source of echo on SIP calls could be ACOUSTIC. The phone sets you are using may not handle this well. In my experience sound quality deteriorates if there is network trouble or congestion on SIP calls I hope this helps. Michael Hi Michael, For sure, i can get echo in the 2 to 4 wire scenario, this is right, but this cant be happen in MY way, only the provider can produce this scenario, my asterisk use zap and isdn, but the echo occure in pure sip calls, in my zap and isdn channels i use the patch from mgernoth, named mg2, great stuff. The second is one echo i already know, one other caller parties use very cheap phones, so the sound of the telephone speaker is not shielded enough to put no sound in the telephone mic - this is not the case with my phones, i use SNOM, they are build to used with VoIP and the best one i know, in my case. I checked the latency and loss between me and my provider this morning again, and i figured out a routing point which lost 3% of my packets, first time for me to see this after one year of working good, i wrote a mail to my provider, and asked him to check this on his own, but i cant imagine that this produce all the echo...must wait, i guess. I tested my Network, good results, tested other VoIP Provider's Server, Result is good to ok. Recap : To minimize echo i can check : Phone (ok), Channels in Asterisk (crossing) (ok), My Network Connections between Phone and Asterisk (ok), Network between Asterisk and Router (ok), Connection, Loss and Latency between Asterisk/Router and my VoiPProvider (waiting..) Any other ways to produce echo in pure *SIP* ! Thanks for your great help ! Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ECHO Cancellation in SIP Calls
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with echocancel=... and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP ! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ECHO Cancellation in SIP Calls
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? The snom phones are pretty decent devices and shouldn't introduce echo. Your latency might be too high between asterisk + voipprovider introducing a delay that is noticed as echo. You are hearing the echo that is introduced on the callers side. As I understand it, when you are calling someone there is no zap involved and thus you can't cancel it with zapata.conf. if you look at voip-info.org [1] you'll find a good explanation why you can't use an echo canceller to cancel that sort of echo. So, check the path between you and the voipprovider, e.g. connection saturation, ping times etc (This is assuming you have a proper lan connection between asterisk/snom) Conrad [1] http://www.voip-info.org/wiki/index.php?page=Asterisk+Echo +Cancellation Hi Conrad, thanks for your help, this is the way i understand it all the time, a year ago i have optimized my Business Lan for VoIP and there is no loss or lag anymore, the Provider seems to be okay, i have pinged him for one day with MTR, no great loss or high ping. Some days ago i have read a Thread about EC-Cancellation in SIP Calls with the zapata.conf and never understood how this could work, thats the beginning of my question ;) In my case, there is no Zap, you are right. So i must start at the beginning and search for a lag... The EC started two or weeks ago after one year of great communication, in this time i updated Asterisk from 1.2.10 to 1.2.12.1 and zaptel from 1.2.7 to 1.2.9 ... I will watch the Quality and the latency next timeThx for your time ! Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 27, Issue 140
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with echocancel=... and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP ! Are you hearing the echo, or is the far end party? I can hear the Echo, the end party never got this problem. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [PROBLEM] Still exist -- DTMF Tones, occures in Asterisk - Channelwide
I don't see anything obviously wrong with your configs. You don't want relaxdtmf. That can cause the problem, not fix it. POST 2 -- got no response Hi Eric, at the begining - Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but my SNOMs working pretty well under RFC2833 , but my cheap Allnet´s cannot handle dtmf unless it is in INFO Mode :( I have now set the Allnets to RFC2833, too, but the DTMF Tones are still there :( I have tried to set the relaxdtmf - but this dont solve the problem, too. I cant understand why it happens on ALL kind of channels, if it would only happen on the SNOMs or something else, i could try to change to INFO or something like this. Have i said that i have updated my Asterisk with the NOTIFY Patch for SNOM Phones ? The configs are under construction, till now :-) Its Asterisk ;) Anybody any Idea ? Need Help...this is ugly..! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [PROBLEM] Still exist -- DTMF Tones, occures in Asterisk - Channelwide
I don't see anything obviously wrong with your configs. You don't want relaxdtmf. That can cause the problem, not fix it. Hi Eric, at the begining - Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but my SNOMs working pretty well under RFC2833 , but my cheap Allnet´s cannot handle dtmf unless it is in INFO Mode :( But, i think, that cannot the problem. I have searched the Internet for talkoff but can only find the relaxdtmf option to increase it. I cant understand why it happens on ALL kind of channels, if it would only happen on the SNOMs or something else, i could try INFO or something like this. This problem exists - i guess few weeks before or after settling up from * 1.2.0 to * 1.2.5... The configs are under construction, till now :-) Its Asterisk ;) Any Idea ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [PROBLEM] Still exist -- DTMF Tones occures in Asterisk - Channelwide
Stefan Agethen schrieb: I got a Problem with DTMF occuring channelwide in my SIP/ZAPPhones. I want to post the Solution for this, so others can find it and help themself. [...] I have updated my Kernel and the first day it seems like my problem was solved, but its not. Problem : DTMF Tones occuring on every kind of channel (SIP/ZAP/mISDN) randomly in the middle of a conversation. Cant reproduce the Situation, but i can log them with DTMF Log of Asterisk (logger.conf) : May 10 15:30:19 DTMF[25551] channel.c: SIP/10-e68c : 1 May 10 15:31:00 DTMF[25551] channel.c: SIP/10-e68c : 1 May 10 15:31:50 DTMF[25551] channel.c: SIP/10-e68c : 1 May 10 15:32:25 DTMF[25551] channel.c: SIP/10-e68c : 1 May 10 17:00:33 DTMF[26110] channel.c: SIP/10-6bcf : 5 May 10 17:06:06 DTMF[26174] channel.c: SIP/30-bcc1 : 5 May 11 06:10:06 DTMF[3633] channel.c: SIP/50-e11c : 9 May 11 08:09:41 DTMF[4853] channel.c: SIP/50-a9b8 : A May 11 08:18:13 DTMF[4945] channel.c: SIP/20-8c2f : C May 11 08:19:46 DTMF[4999] channel.c: SIP/50-42fb : 9 The whole day long this problem appears and in the baddest case Asterisk get a # or * and some features get started like MOH and so on. I have disabled this features in the most cases, but still i get the Tones while talking :-( Please help ! Cant find any solution Internetwide ! I use DTMF RFC2833 for snom and ZAPs - INFO for my Allnet-Sips. Will give any info if someone try to help ! Best wishes, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [PROBLEM] Still exist -- DTMF Tones occures in Asterisk - Channelwide
I got a Problem with DTMF occuring channelwide in my SIP/ZAPPhones. I have updated my Kernel and the first day it seems like my problem was solved, but its not. The problem is called talkoff. Search the mailing list archives. Also post your sip.conf and zaptel.conf (sans passwords, of course) Thx, now got my problem a name! Here is my sip.conf and zaptel.conf : SIP--- [general] port = 5060 bindaddr = 0.0.0.0 externhost=agethen.no-ip.org localnet=10.0.0.0/255.255.255.0 srvlookup=yes fromdomain=10.0.0.60 language=de tos=0x18 nat=yes disallow=all allow=alaw allow=g726 defaultexpirey=3600 maxexpirey=3600 REGISTER-STATEMENTS register=xx:[EMAIL PROTECTED]/xx [pbx1] ;VOIP Provider bindport=5060 username=xx type=peer secret=xx qualify=no insecure=very host=sip.pbx-network.de fromuser=xx fromdomain=pbx-network.de context=voip_eingehend canreinvite=no dtmfmode=info tos=0x18 [...] Now there are many SIP Phone and VOIP Provider Contexts - too much for here, i think. ZAPTEL--- fxoks=1 fxoks=2 fxoks=3 fxoks=4 loadzone= de defaultzone= de Can RelaxDTMF do something for me ? Best wishes, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [SOLUTION] DTMF Tones occures in Asterisk
I got a Problem with DTMF occuring channelwide in my SIP/ZAPPhones. I want to post the Solution for this, so others can find it and help themself. Problem : DTMF Tones like 9 * # been sent randomly in the middle of my conversations. That was distracting and sometimes a bad DTMF Tone like * or # could do more than i believe ;) I cant figure out why there a occure and how. Tracing : Activate the LOG for DTMF Tones in Asterisk itself. (logger.conf) Solution : Check your Kernel Version, i used 2.6.8-2-386 with Debian Sarge, i have updated Debian with the backports in the sources.list , easy, safe and fast, recompiled the whole asterisk and addons and i cant believe, no dtmf anymore ! If you are in need - post back. Best Wishes, Stefan from Germany in school english! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Tones within my Asterisk on all type of Channels
Hi, i am using Asterisk 1.2.6 with Debian Kernel 2.6.8-2-386 and up 2 date Zaptel, Libpri and so on. My Hardware : 4 x SNOM 320 One Wildcard TDM40B 3 x Allnet 7950 1 x Fritz!Box 7050 - working as a ATA I use Asterisk in my Business stable since last year, the problem occurs since 2 or 3 months. I have detected some DTMF Tones within my conversations and turned on a DTMF Logfile to get more infos about this. The LOG reports Tones - they looks like appearing randomly on any type of channel (mISDN, ZAP, SIP) and i cant find any reason why. Here is a part of the LOG : Apr 13 10:18:59 DTMF[5007] channel.c: SIP/30-1352 : * Apr 13 10:39:52 DTMF[5165] channel.c: SIP/20-5073 : # [...] Apr 15 07:50:53 DTMF[14182] channel.c: Zap/1-1 : A Apr 15 07:51:44 DTMF[14182] channel.c: Zap/1-1 : A Apr 15 13:04:36 DTMF[14909] channel.c: SIP/50-7aa2 : 4 Apr 16 20:20:11 DTMF[19677] channel.c: mISDN/1-u61 : 9 [...] Apr 18 07:33:32 DTMF[26583] channel.c: Zap/1-1 : 5 Apr 18 07:38:50 DTMF[26663] channel.c: SIP/50-21c4 : 9 [...] As you can see, this happens on any type of available channel, the user can hear the tone and in the baddest case the * appears and the function to transfer in the DIAL is active (tT).. Can you give me a tip to get this problem solved or locate it ? Best Greets from Germany, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
