Re: [asterisk-users] No ringback heard
On Thu, Aug 25, 2016 at 2:14 PM, Saint Michaelwrote: > I dial two destination like this > > Dial(PJSIP/endpoint1/sip:${EXTEN}@${IPA}/endpoint1/sip:${EXTEN}@ > ${IPB}) > > But I need the audio from one of them to be heard by the caller. > None gets heard. I switch the order but nothing. > How I get the audio for one in particular? > You cannot. In the general case, forked dials like that cannot pass any early media back to the caller because you would need to mix any early media from the two (or more) outgoing channels. In addition, any mixed audio would be confusing. Imaging hearing "an all circuits are busy" recording while at the same time hearing overlapping ringback tones from several other channels. In specific cases, you are talking about a new feature which requires a code change. If what you want is ringback with a recording interspersed at intervals, you can create a music-on-hold class and have the caller hear that instead. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No ringback heard
I dial two destination like this Dial(PJSIP/endpoint1/sip:${EXTEN}@${IPA}/endpoint1/sip:${EXTEN}@ ${IPB}) But I need the audio from one of them to be heard by the caller. None gets heard. I switch the order but nothing. How I get the audio for one in particular? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no ringback tone on outgoing call PRI line
Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP-PRI mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no ringback tone on outgoing call PRI line
https://issues.asterisk.org/view.php?id=18868 -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 8 May 2011 11:43:41 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] no ringback tone on outgoing call PRI line -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 7 April 2011 23:04, Douglas Mortensen d...@impalanetworks.com wrote: Steve. Thanks for the insight. I won't pretend to know what early-audio is, but I guess I'm about to find out :-). Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that system must be running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is there a significant difference between 1.2/1.4 1.6 in this scenario? Thanks a million!! :-) - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: Steve Davies [mailto:davies...@gmail.com] Sent: Thursday, April 07, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve Early audio is audio that is sent before the call is answered, usually in the form of a custom ring-tone or perhaps a cannot connect, try later message. Some systems do not support it as it can be abused to communicate at least basic information for free. We had a problem with this when connecting Asterisk 1.2 to Asterisk 1.6 via IAX. A 1.2 SIP system will automatically switch into early audio if it sees an early audio frame. 1.6 defaults to not doing this, but there is a parameter to re-enable it. In this case we solved the problem by upgrading to 1.6 everywhere :) Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No ringback even though progressinband=yes is set
Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 4/7/2011 11:02 AM, Douglas Mortensen wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you are referring to a ringback tone when they first dial your system, meaning that they immediately hear your IVR when they dial your PBX's number, it's because that's how it's supposed to work. Unless you tell your PBX to use the Ringing() app and wait for a period of time, Asterisk normally picks up at the beginning of the IVR (since the first thing you have to do to send audio via Background or Playback is issue the command Answer() to start sending actual audio. (Note: The Ringing app just signals RINGING to the remote party) -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
I have inbound calls going directly to a ring group. When callers call in, they (the callers) hear complete silence even though the phones that are part of the ring group ARE ringing properly. Employees can answer the calls when their phones ring, and everything works fine. The problem is simply that the external caller never hears any ringing. Even if the SIP phones in the ring group ring for 5 rings, it is total silence even though there is ringing going on inside of the office. I'm pretty sure it is a ringback issue. I'm going to try to turn on SIP debugging see what I can figure out that way. I do appreciate your help. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: Sherwood McGowan [mailto:sherwood.mcgo...@gmail.com] Sent: Thursday, April 07, 2011 12:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 4/7/2011 11:02 AM, Douglas Mortensen wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you are referring to a ringback tone when they first dial your system, meaning that they immediately hear your IVR when they dial your PBX's number, it's because that's how it's supposed to work. Unless you tell your PBX to use the Ringing() app and wait for a period of time, Asterisk normally picks up at the beginning of the IVR (since the first thing you have to do to send audio via Background or Playback is issue the command Answer() to start sending actual audio. (Note: The Ringing app just signals RINGING to the remote party) -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
Steve. Thanks for the insight. I won't pretend to know what early-audio is, but I guess I'm about to find out :-). Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that system must be running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is there a significant difference between 1.2/1.4 1.6 in this scenario? Thanks a million!! :-) - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: Steve Davies [mailto:davies...@gmail.com] Sent: Thursday, April 07, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 4/7/2011 4:54 PM, Douglas Mortensen wrote: I have inbound calls going directly to a ring group. When callers call in, they (the callers) hear complete silence even though the phones that are part of the ring group ARE ringing properly. Employees can answer the calls when their phones ring, and everything works fine. The problem is simply that the external caller never hears any ringing. Even if the SIP phones in the ring group ring for 5 rings, it is total silence even though there is ringing going on inside of the office. I'm pretty sure it is a ringback issue. I'm going to try to turn on SIP debugging see what I can figure out that way. I do appreciate your help. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 If you're using an interface (I believe you said AsteriskNOW), you might want to check the Dial Options...Make sure that 'r' is one of the options. The reason you're not hearing ringing is probably due to Asterisk not sending a RINGING signal. If you have 'r' defined in the dial options in your interface, then AsteriskNOW is probably using a Dial command that is NOT using your global dial options. -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fake ringback tone
Hi: Iam not using the 'r' option in my dial plan ,here what i have in my dial plan: [gw]exten = _70.,1,Dial,SIP/grands/${EXTEN} Date: Fri, 9 Jan 2009 16:25:41 -0500 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] fake ringback tone On Fri, Jan 9, 2009 at 3:57 PM, wassim Darwish wassim...@hotmail.com wrote: hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. If you are using the r option in your Dial statement, remove it. That generates fake ringing. In FreePBX, that option is under the General settings, if plain jane Asterisk, just remove the r in your dial line. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009___ -- 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
Re: [asterisk-users] fake ringback tone
wassim Darwish schrieb: Iam not using the 'r' option in my dial plan ,here what i have in my dial plan: Hint: Don't remove the line breaks: [gw]exten = _70.,1,Dial,SIP/grands/${EXTEN} Date: Fri, 9 Jan 2009 16:25:41 -0500 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] fake ringback tone On Fri, Jan 9, 2009 at 3:57 PM, wassim Darwish wassim...@hotmail.com wrote: hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. If you are using the r option in your Dial statement, remove it. That generates fake ringing. In FreePBX, that option is under the General settings, if plain jane Asterisk, just remove the r in your dial line. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list T o UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] fake ringback tone
hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. _ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012009___ -- 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
Re: [asterisk-users] fake ringback tone
On Fri, Jan 9, 2009 at 3:57 PM, wassim Darwish wassim...@hotmail.com wrote: hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. If you are using the r option in your Dial statement, remove it. That generates fake ringing. In FreePBX, that option is under the General settings, if plain jane Asterisk, just remove the r in your dial line. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Providing Ringback
Hello, We've had this problem happen twice with retail customers already and still have no solution. Basically there are times when customers can't get any ring at all. It happens that they call our switch and even though we are receiving ring from the carrier they hear no ring. We have even put a fake-ring(with Rr) back at their request and they are unable to get this ring either. The first time it happened was with a customer running a Cisco switch, now more recently we have a customer with VoipSwitch that gets no ring. Our other customers receive the ring from the carrier fine. Has anyone experienced this before and if so how did you solve it? Regards, Igor Hernandez Escape Communications. ___ -- 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
Re: [asterisk-users] Providing Ringback
Hi Igor, We had an interconnect with a carrier that generated early media for progress indications but the carrier's switch, in this case a Cerpack, would only start sending the RTP for the early media AFTER it received an RTP packet from the Asterisk end. Completely stupid behaviour since early media is generally only one way but that's what it did. We worked around it by recording 200ms of silence and playing that back to the carrier's Cerpack with the Background command whenever we received an incoming call. This got two way RTP set up and allowed the progress tones to be correctly passed through to the user. [noringback] exten = _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n) exten = _X.,2,Goto(incoming, ${EXTEN}, 1) Regards, Greyman. ___ -- 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
Re: [asterisk-users] Providing Ringback
Thanks a lot Grey. I'll look into it. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com Grey Man wrote: Hi Igor, We had an interconnect with a carrier that generated early media for progress indications but the carrier's switch, in this case a Cerpack, would only start sending the RTP for the early media AFTER it received an RTP packet from the Asterisk end. Completely stupid behaviour since early media is generally only one way but that's what it did. We worked around it by recording 200ms of silence and playing that back to the carrier's Cerpack with the Background command whenever we received an incoming call. This got two way RTP set up and allowed the progress tones to be correctly passed through to the user. [noringback] exten = _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n) exten = _X.,2,Goto(incoming, ${EXTEN}, 1) Regards, Greyman. ___ -- 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 ___ -- 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
Re: [asterisk-users] Bad ringback tone on zap channel
The ringback is coming from the Zap channel, since that's the destination of the call. Therefore, the bad ring is more likely to be coming from the remote end. What type of line are you making the call to? Analogue? E1/T1? If it's analogue, I'd be guessing you have a faulty PSTN line. James Lamanna wrote: Hmm ok. This was a call from a SIP phone registered with Asterisk outbound on a Zap trunk. So would Asterisk or the phone be generating the ringback tone in that case? It also happens very intermittently (maybe 1 in 10 calls at most...) -- James Rob Hillis wrote: In my experience, the ringback you get over a zap channel (be it analogue or digital) is generated by the remote end, /not/ Zaptel. The ringback you get over a SIP or IAX2 channel is often generated by either Asterisk or the SIP/IAX2 device you're calling from. James Lamanna wrote: Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. ___ -- 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 !DSPAM:484b09be67791587961402! ___ -- 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
Re: [asterisk-users] Bad ringback tone on zap channel
In my experience, the ringback you get over a zap channel (be it analogue or digital) is generated by the remote end, /not/ Zaptel. The ringback you get over a SIP or IAX2 channel is often generated by either Asterisk or the SIP/IAX2 device you're calling from. James Lamanna wrote: Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks. ___ -- 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 !DSPAM:484975fe67791857117240! ___ -- 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
Re: [asterisk-users] Bad ringback tone on zap channel
Hmm ok. This was a call from a SIP phone registered with Asterisk outbound on a Zap trunk. So would Asterisk or the phone be generating the ringback tone in that case? It also happens very intermittently (maybe 1 in 10 calls at most...) -- James Rob Hillis wrote: In my experience, the ringback you get over a zap channel (be it analogue or digital) is generated by the remote end, /not/ Zaptel. The ringback you get over a SIP or IAX2 channel is often generated by either Asterisk or the SIP/IAX2 device you're calling from. James Lamanna wrote: Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. ___ -- 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] Bad ringback tone on zap channel
Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks. ___ -- 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] no ringback from SIP server when originating call
I have an application that uses the Asterisk Management Interface to bridge two calls using the Originate command with Dial as the action. Using one SIP server, there is no ringback on the second leg of the call. The first person is called, answers, and hears silence until the second person picks up, even though the second person's phone is ringing. When the call goes to another SIP gateway, ringback works fine. From SIP traces I found that the one that works returns 180 ringing to Asterisk and the one that doesn't work returns 100 trying followed by 183 session progress. It is my understanding that 180 ringing causes ringback to be generated by the callee, while 183 means that the caller has early media and will send ringback through RTP. Anyone have any idea why I wouldn't get ringback in this case? Should Asterisk be passing through the early media to the first caller even though the second caller has not answered? I am not using the r option to the Dial command. I have tried it both on and off and get no ringback in either case. I have also tried variations of the progressinband setting. I have listened to the RTP going from the SIP server to Asterisk and I can hear the ringing in it. It seems like Asterisk isn't sending any audio to the first caller until both parties answer. Thanks, Matthew Boedicker ___ --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
Re: [asterisk-users] MixMonitor RingBack Tone Issue
Hi Jean-Marc, I tried to use mixmonitor and seems that it works good. My problem is about calls after a transfer: it seems that asterisk can completely record a call in one file, only in case of blind transfer. If I make an attended transfer I have 2 or more sound files which are impossible to join. Have you successfully recorded sound files of transfered calls in one file?? TIA Giorgio Incantalupo Jean-Marc Salsa wrote: Indeed, perfect ! Thanks a lot ... JM On 2/17/07, *Trevor Peirce* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jean-Marc Salsa wrote: exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r mailto: SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Yes because you have the r in there, asterisk sends its own ringing. If you want ringing to be heard from the PSTN, you need to leave that option disabled. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor RingBack Tone Issue
Indeed, perfect ! Thanks a lot ... JM On 2/17/07, Trevor Peirce [EMAIL PROTECTED] wrote: Jean-Marc Salsa wrote: exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r mailto:SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Yes because you have the r in there, asterisk sends its own ringing. If you want ringing to be heard from the PSTN, you need to leave that option disabled. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor RingBack Tone Issue
Hi, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN - GW -SIP- Asterisk(Record) -SIP- Softswitch - IP Phone outgoing calls : IP Phone - Softswitch -SIP- Asterisk(Record) -SIP- GW - PSTN Dial plan in Asterisk is quite simple: [record] exten = s,1,Set(CALLFILENAME=${TIMESTAMP}-${UNIQUEID}) exten = s,n,Set(CALLERID(name)=${CALLERID(name)}) exten = s,n,Set(CALLERID(number)=00${CALLERID(number)}) exten = s,n,MixMonitor(${CALLFILENAME}.WAV,b) exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Any ideas why ? How can I bypass this issue ? Thanks, Jean-Marc ___ --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] MixMonitor RingBack Tone Issue
Jean-Marc Salsa wrote: exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r mailto:SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Yes because you have the r in there, asterisk sends its own ringing. If you want ringing to be heard from the PSTN, you need to leave that option disabled. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Ringback, only on 1 SIP provider
Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and International, both via SIP. The problem I am having is that the users dont get ringback (ringing indication) when they dial International numbers, yet it works perfectly when they dial Local numbers. Yet, to test, from a hardphone plugged into Asterisk2, I get ringback, so its not the Interntional provider, it must be the SIP trunk from Asterisk1 to Astrisk2. (ringback) NationalProvider | SIP| | H323 SIP | SIP (no ringback) Users phones - CCM 4.1 Asterisk1-Asterisk2-InternationalProvider | | ZAP hardphone Here is the sip.conf from Asterisk1. [N_G] type=friend host=10.255.255.1 username=N_G secret=N_G disallow=all allow=g729 canreinvite=no qualify=yes progressinband=yes (tried this yes/no/never, made no difference) When I call goes from Asterisk1 to Asterisk2, I get the 'making progress passing it to xxx', but I dont hear ringing, then the person answers. -- thanks, Yusuf ___ --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] Callback/ringback
Yehavi Bourvine +972-8-9489444 wrote: Enclosed bellow is the fragment from extenstions.conf which does two things: *41 - Does the ring-back staff. *42 - Calls back the last one who called you. Regards, __Yehavi: That's a very nice little script. -- Warm Regards, Lee ___ --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] Callback/ringback
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Excellent little script. Thanks, Yehavi. Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. -Original Message- From: Yehavi Bourvine +972-8-9489444 [mailto:[EMAIL PROTECTED] Sent: 18 January 2007 07:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Callback/ringback Enclosed bellow is the fragment from extenstions.conf which does two things: *41 - Does the ring-back staff. *42 - Calls back the last one who called you. Regards, __Yehavi: ; regular local extensions: ; The flow is: If not available or no answer send to mailbox if exists, ; send busy if no mailbox. Same for busy. ; We try to avoid the n+101 rule whenever possible, but it is not always ; possible as HasVoiceMailbox() does only n+101 jump. exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled. exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller. ; Save the caller number at the called extension for *42 usage. exten = _999XX,n,Set(DB(${To}/LastCaller)=${From}) ; Where we called for *41 exten = _999XX,n,Set(DB(${From}/LastCalled)=${To}) ; Now dial the extension. exten = _999XX,n,Dial(SIP/${EXTEN},20,) ; Dial the phone for 20 seconds. ; No answer or busy exten = _999XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure mode exten = _999XX,n,Hangup() ; Just to be sure... ; No answer: exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox? exten = s-NOANSWER,n,Busy(); No maibox = play busy. exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call to there ; Busy: exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox? exten = s-BUSY,n,Busy(); No maibox = play busy. exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to there ; Unavailable channel - act as busy: exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1); ; Called here when the call is successfull and the user hanged the phone. ; Check whether the user has a waiting callback queued on him/her exten = h,1,NoOp(${From} ${To} ${EXTEN}) exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us exten = h,3,NoOp(${From} ${tmp}) exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us? exten = h,5,DBdel(${From}/CallBack); And delete it... ; Create the callfile and then move it to the spool directory to make the call. exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To}) exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To}) exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To}) exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\ /tmp/test.tmp${To}) exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/) exten = h,103,NoOp(Nothing to call) ; *42: Get the last number who called us, say it and call it. exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller}) exten = *42,n,SayDigits(${tmp}) exten = *42,n,Goto(${tmp},1) ; *41: Camp on the last extension dialled exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)}) exten = *41,n,SayDigits(${tmp}) ; Save it so when the other side hangs it will see it and dial us. exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)}) exten = *41,n,Hangup() ___ --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] Callback/ringback
Richard Soderblom wrote: Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is busy or unavailable then Asterisk should inform the first user that the number they dialed is busy and hangup the call. Once the second caller is available again then Asterisk should initiate a call back to both the users and connect them. Any ideas on how to achieve this will be appreciated. Richard, That shouldn't be too difficult to do. I recently wrote an agi binary that does nag calling for me which I think is related to what you want to do except that I am doing more calling out of the system. Maybe deadagi could work? Here is it's use in a AEL macro I'm working on: http://www.datatrakpos.com/pos/datatalk/images/nagcall.htm The AGI (nagcall) simply takes some parameters and uses them to create a .call file. See: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out There is even an example on the page above that shows using the linux touch command to schedule the call to take place at a later time, although I have not successfully done this yet...still trying. The biggest difference is that you will need a way to monitor the called extension to trigger a call back to the original caller using maybe deadagi or .call files? http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+API I'm pretty new to asterisk myself so there may be (probably are) other ways to do this, but this is where I would start poking around. -- Warm Regards, Lee ___ --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] Callback/ringback
Enclosed bellow is the fragment from extenstions.conf which does two things: *41 - Does the ring-back staff. *42 - Calls back the last one who called you. Regards, __Yehavi: ; regular local extensions: ; The flow is: If not available or no answer send to mailbox if exists, ; send busy if no mailbox. Same for busy. ; We try to avoid the n+101 rule whenever possible, but it is not always ; possible as HasVoiceMailbox() does only n+101 jump. exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled. exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller. ; Save the caller number at the called extension for *42 usage. exten = _999XX,n,Set(DB(${To}/LastCaller)=${From}) ; Where we called for *41 exten = _999XX,n,Set(DB(${From}/LastCalled)=${To}) ; Now dial the extension. exten = _999XX,n,Dial(SIP/${EXTEN},20,) ; Dial the phone for 20 seconds. ; No answer or busy exten = _999XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure mode exten = _999XX,n,Hangup() ; Just to be sure... ; No answer: exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox? exten = s-NOANSWER,n,Busy(); No maibox = play busy. exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call to there ; Busy: exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox? exten = s-BUSY,n,Busy(); No maibox = play busy. exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to there ; Unavailable channel - act as busy: exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1); ; Called here when the call is successfull and the user hanged the phone. ; Check whether the user has a waiting callback queued on him/her exten = h,1,NoOp(${From} ${To} ${EXTEN}) exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us exten = h,3,NoOp(${From} ${tmp}) exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us? exten = h,5,DBdel(${From}/CallBack); And delete it... ; Create the callfile and then move it to the spool directory to make the call. exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To}) exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To}) exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To}) exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\ /tmp/test.tmp${To}) exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/) exten = h,103,NoOp(Nothing to call) ; *42: Get the last number who called us, say it and call it. exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller}) exten = *42,n,SayDigits(${tmp}) exten = *42,n,Goto(${tmp},1) ; *41: Camp on the last extension dialled exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)}) exten = *41,n,SayDigits(${tmp}) ; Save it so when the other side hangs it will see it and dial us. exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)}) exten = *41,n,Hangup() ___ --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] Callback/ringback
Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is busy or unavailable then Asterisk should inform the first user that the number they dialed is busy and hangup the call. Once the second caller is available again then Asterisk should initiate a call back to both the users and connect them. Any ideas on how to achieve this will be appreciated. Thanks, Richard . ___ --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] SOLVED: ringback on box with E1 and premicell
Hi, I had been struggling with this, and I thought I will post the solution. I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. This is true *only* when in zapata.conf with answeronpolarityswitch=yes. The ZAP device (a premicell), sends polarityswitches when the call starts and when the call ends. in zapata.conf with answeronpolarityswitch=yes then when the phone starts to ring, you dont hear it ring, only when the person answers the phone do you start to hear him talk. So therefore I do not hear the phone ring when answeronpolarityswitch=yes SOLUTION: in zaptel.conf loadzone=za in zapata.conf callprogress=yes progzone=za priindication=outofband -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Outgoing Ringback Indications IAX vs. SIP
On 17 Apr 2006, at 00:30, Steve Feinstein wrote: Actually it makes no difference. I tried it in an attempt to get something to happen. Thanks, -Steve Eric ManxPower Wieling wrote: What happens if you remove the r option? r is almost NEVER useful. Steve Feinstein wrote: I've been pulling my hair out over this one trying to understand it. If you have a very simple extension: exten = 1,n,Dial(IAX2/Steve|24|r) Everything I've seen says this should tell the IAX phone (our own iaxclient based one) to make a ringing sound, or asterisk should make the ringback indication itself if it determines that the channel can't do it for itself. But you can dial this extension all day and you never hear a ringback indication. Dial it from a SIP softphone and you do. If you change the default country in the indications.conf, the SIP phone will change the way the ring sounds. IAX, still nothing. You can use PlayTones(ring) in the dialplan before the Dial(), and it seems to behave ok. Playing the appropriate ring indication until the call is answered. But it seems like the behavior is inconsistent with IAX vs. SIP. Is this by design? All the IAX soft phones I've tried are based on the same iaxclient libs, so it's hard to know if it's the phone or asterisk that's not behaving right. Has anyone used an iax hard phone, some other IAX device/software, and does it exhibit the same behavior? Or is this a problem with the iax code not being telling asterisk that IAX phones need to have their indications faked. Any ideas about what's going on would be most gratefully appreciated. I don't know the IAXclient libs, but an IAX client is supposed to send a RINGING packet back after it accepts a call to notify the other end it should generate ringback for the user. The protocol allows it to go straight to ANSWER, or send a PROCEEDING if it hasn't reached the end- point yet. Is your client sending a RINGING packet at the right moment ? Is there a call you should make (after accept but before answer) to get it to send RINGING? Tim Panton [EMAIL PROTECTED] ___ --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] Outgoing Ringback Indications IAX vs. SIP
What happens if you remove the r option? r is almost NEVER useful. Steve Feinstein wrote: I've been pulling my hair out over this one trying to understand it. If you have a very simple extension: exten = 1,n,Dial(IAX2/Steve|24|r) Everything I've seen says this should tell the IAX phone (our own iaxclient based one) to make a ringing sound, or asterisk should make the ringback indication itself if it determines that the channel can't do it for itself. But you can dial this extension all day and you never hear a ringback indication. Dial it from a SIP softphone and you do. If you change the default country in the indications.conf, the SIP phone will change the way the ring sounds. IAX, still nothing. You can use PlayTones(ring) in the dialplan before the Dial(), and it seems to behave ok. Playing the appropriate ring indication until the call is answered. But it seems like the behavior is inconsistent with IAX vs. SIP. Is this by design? All the IAX soft phones I've tried are based on the same iaxclient libs, so it's hard to know if it's the phone or asterisk that's not behaving right. Has anyone used an iax hard phone, some other IAX device/software, and does it exhibit the same behavior? Or is this a problem with the iax code not being telling asterisk that IAX phones need to have their indications faked. Any ideas about what's going on would be most gratefully appreciated. ___ --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] Outgoing Ringback Indications IAX vs. SIP
Actually it makes no difference. I tried it in an attempt to get something to happen. Thanks, -Steve Eric ManxPower Wieling wrote: What happens if you remove the r option? r is almost NEVER useful. Steve Feinstein wrote: I've been pulling my hair out over this one trying to understand it. If you have a very simple extension: exten = 1,n,Dial(IAX2/Steve|24|r) Everything I've seen says this should tell the IAX phone (our own iaxclient based one) to make a ringing sound, or asterisk should make the ringback indication itself if it determines that the channel can't do it for itself. But you can dial this extension all day and you never hear a ringback indication. Dial it from a SIP softphone and you do. If you change the default country in the indications.conf, the SIP phone will change the way the ring sounds. IAX, still nothing. You can use PlayTones(ring) in the dialplan before the Dial(), and it seems to behave ok. Playing the appropriate ring indication until the call is answered. But it seems like the behavior is inconsistent with IAX vs. SIP. Is this by design? All the IAX soft phones I've tried are based on the same iaxclient libs, so it's hard to know if it's the phone or asterisk that's not behaving right. Has anyone used an iax hard phone, some other IAX device/software, and does it exhibit the same behavior? Or is this a problem with the iax code not being telling asterisk that IAX phones need to have their indications faked. Any ideas about what's going on would be most gratefully appreciated. ___ --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 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/313 - Release Date: 4/15/2006 ___ --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] Outgoing Ringback Indications IAX vs. SIP
I've been pulling my hair out over this one trying to understand it. If you have a very simple extension: exten = 1,n,Dial(IAX2/Steve|24|r) Everything I've seen says this should tell the IAX phone (our own iaxclient based one) to make a ringing sound, or asterisk should make the ringback indication itself if it determines that the channel can't do it for itself. But you can dial this extension all day and you never hear a ringback indication. Dial it from a SIP softphone and you do. If you change the default country in the indications.conf, the SIP phone will change the way the ring sounds. IAX, still nothing. You can use PlayTones(ring) in the dialplan before the Dial(), and it seems to behave ok. Playing the appropriate ring indication until the call is answered. But it seems like the behavior is inconsistent with IAX vs. SIP. Is this by design? All the IAX soft phones I've tried are based on the same iaxclient libs, so it's hard to know if it's the phone or asterisk that's not behaving right. Has anyone used an iax hard phone, some other IAX device/software, and does it exhibit the same behavior? Or is this a problem with the iax code not being telling asterisk that IAX phones need to have their indications faked. Any ideas about what's going on would be most gratefully appreciated. -Steve Feinstein (asterisk 1.2.7.1 btw) GatherWorks, Inc. begin:vcard fn:Steve Feinstein n:Feinstein;Steve org:GatherWorks Inc. adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA email;internet:[EMAIL PROTECTED] tel;work:+1 (603) 672-1472 x-mozilla-html:TRUE url:http://www.gatherworks.com/ version:2.1 end:vcard ___ --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] IAXy Ringback Issues
Here is a very strange problem Im running into with all of the IAXys were testing in regards to ringback when placing an outgoing call. First, let me describe the setup. We have a production Asterisk 1.0.9 box running that uses SIP trunks to connect to various Cisco voice gateways. Those Cisco gateways then connect to internal switches (DMS-100s) via PRIs. The analog phone switch it connects to uses 5 digit dialing for internal numbers, and 7 or 10 digit dialing for external and LD numbers. Therefore, when I am dialing from an IAXy through Asterisk to another analog internal extension, I am just using 6 digits. I have the dial plan setup to match the internal extensions as their 10, 7, or 5 digit numbers. If, for some reason, the person dials the full 7 or 10 digits to dial an internal neighbor, it just strips the unnecessary numbers and sends it to the trunk. Now heres the problem with the IAXy. If I dial an internal extension from the IAXy as X (5 digit), the destination rings and the call completes, but I DO NOT hear ringback tone. If I dial an internal extensions as XXX- (7 digit) or XXX-XXX-X (10 digit), the call completes and I DO get ringback tone. Ive looked in the logs, and the dial plan is working correctly, whether the person dials 5, 7 or 10 digits, asterisk is only sending 5 digits to the trunk to complete the call. Also, I thought it may be the IAX driver causing problems, but we have a number of Chinese made phones with ATCOM chips that use the IAX protocol, none of which are experiencing the same ringback tone issues. Ive also tried the beta version of the IAXy firmware Does this mean the IAXy has some sort of dial plan that controls when it creates ringback tone? Any help would be appreciated Thanks, Scott Miller ___ --Bandwidth and Colocation sponsored by Easynews.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] No ringback tone generated by Asterisk with OH323 connections
I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=3 udpStart=20001 udpEnd=3 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c21.15.3-4 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.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] No ringback tone generated by Asterisk with OH323connections
are you giving answer()? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=3 udpStart=20001 udpEnd=3 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c21.15.3-4 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.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] No ringback tone generated by Asterisk with OH323connections
No, I wasn't. I can't believe I made that stupid mistake. It started working after I added the call to Answer(). Thanks for your help. On Friday 30 September 2005 11:53, Brian C. Fertig wrote: are you giving answer()? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=3 udpStart=20001 udpEnd=3 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk 1.0.9.dfsg-5 asterisk-oh3230.6.6pre3-4 libopenh323-1.15.3c2 1.15.3-4 -- Juan Jose Comellas ([EMAIL PROTECTED]) -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.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] NO ringback tone for VOIP call to another SIP server
All, I found that there is no ringback to the caller (a-party) for VoIP call but when I make call to registered user, I can hear the ringback tone. Beloware the debug log for the two cases: I wonder if anyone who can tell me why? Thanks. Raymond Case 1: no ringback to the caller (a-party) for outbond VoIP call to another SIP server Apr 26 07:04:09 VERBOSE[2607]: -- Executing Dial("SIP/30511694-abfa", "SIP/[EMAIL PROTECTED]") in new stackApr 26 07:04:09 DEBUG[2607]: Outgoing Call for 99740185293137656Apr 26 07:04:09 DEBUG[2607]: 99740185293137656 is not a local userApr 26 07:04:09 VERBOSE[2607]: -- Called [EMAIL PROTECTED]Apr 26 07:04:09 DEBUG[2607]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:04:13 DEBUG[2607]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:04:13 VERBOSE[2607]: -- SIP/192.168.11.194-8dc7 is making progress passing it to SIP/30511694-abfaApr 26 07:04:13 DEBUG[2607]: Ooh, format changed from unknown to ulawApr 26 07:04:13 DEBUG[2607]: Auto destroying call '[EMAIL PROTECTED]'Apr 26 07:04:13 DEBUG[2607]: RTP NAT: Using address 192.168.19.241:64868Apr 26 07:04:13 DEBUG[2607]: Oooh, format changed to 8Apr 26 07:04:13 DEBUG[2607]: Ooh, format changed from unknown to ulawApr 26 07:04:13 DEBUG[2607]: Ooh, format changed from ulaw to alawApr 26 07:04:15 NOTICE[2607]: RFC3389 support incomplete. Turn off on client if possibleApr 26 07:04:32 DEBUG[2607]: update_user_counter(99740185293137656) - decrement outUse counterApr 26 07:04:32 DEBUG[2607]: 99740185293137656 is not a local userApr 26 07:04:32 DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.Apr 26 07:04:32 VERBOSE[2607]: == Spawn extension (siptest02, 85293137656, 1) exited non-zero on 'SIP/30511694-abfa'Apr 26 07:04:32 VERBOSE[2607]: -- Executing Hangup("SIP/30511694-abfa", "") in new stackApr 26 07:04:32 VERBOSE[2607]: == Spawn extension (siptest02, h, 1) exited non-zero on 'SIP/30511694-abfa'Apr 26 07:04:32 DEBUG[2607]: cdr_mysql: inserting a CDR record.Apr 26 07:04:32 DEBUG[2607]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-04-26 07:04:09','\"cisco 7960\" 30511694','30511694','85293137656','siptest02', 'SIP/30511694-abfa','SIP/192.168.11.194-8dc7','Hangup','',23,0,'NO ANSWER',3,'')Apr 26 07:04:32 DEBUG[2607]: update_user_counter(30511694) - decrement inUse counterApr 26 07:04:32 DEBUG[2607]: Acked pending invite 102Apr 26 07:04:32 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: FoundApr 26 07:04:32 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: FoundApr 26 07:04:32 DEBUG[2607]: 99740185293137656 is not a local userApr 26 07:04:32 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Found Case 2: When I make call to registered user, I can hear the ringback tone: Apr 26 07:05:49 DEBUG[2607]: Auto destroying call '[EMAIL PROTECTED]'Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4Apr 26 07:05:50 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: FoundApr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4Apr 26 07:05:50 DEBUG[2607]: Check for res for 30511694Apr 26 07:05:50 DEBUG[2607]: Call from user '30511694' is 1 out of 0Apr 26 07:05:50 DEBUG[2607]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060Apr 26 07:05:50 VERBOSE[2607]: -- Executing Dial("SIP/30511694-581e", "SIP/30511690|20|tr") in new stackApr 26 07:05:50 DEBUG[2607]: SIMPLE DIAL (NO URL)Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4Apr 26 07:05:50 DEBUG[2607]: Outgoing Call for 30511690Apr 26 07:05:50 DEBUG[2607]: Call from user '30511690' is 1 out of 0Apr 26 07:05:50 VERBOSE[2607]: -- Called 30511690Apr 26 07:05:50 DEBUG[2607]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:05:50 DEBUG[2607]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:05:50 VERBOSE[2607]: -- SIP/30511690-adb1 is ringingApr 26 07:06:00 DEBUG[2607]: update_user_counter(30511690) - decrement outUse counterApr 26 07:06:00 DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.Apr 26 07:06:00 VERBOSE[2607]: == Spawn extension (siptest02, 1690, 1) exited non-zero on 'SIP/30511694-581e'Apr 26 07:06:00 VERBOSE[2607]: -- Executing Hangup("SIP/30511694-581e", "") in new stackApr 26 07:06:00 VERBOSE[2607]: == Spawn extension (siptest02, h, 1) exited non-zero on 'SIP/30511694-581e'Apr 26 07:06:00 DEBUG[2607]: cdr_mysql: inserting a CDR record.Apr 26 07:06:00 DEBUG[2607]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES
Re: [Asterisk-Users] No ringback over IAX - LiveVoip
Found this info on their website: http://www.livevoip.com/index.php?subject=2content=networkStatus LiveVoip Operations Staff DTMF - Ringback Issues Currently, Asterisk is using the timing of the input stream to reproduce the output stream. This means that when no RTP streams are being sent from the peer Endpoint Gateway, Asterisk is unable to generate audio. This approach or limitation leads to one way speech conditions. Plus - Some devices don't generate audio until the answer supervision is received from the called. For all these scenarios, no ringback can be presented to the calling party. In cases where the endpoints are using silence compression, the audio from asterisk is chopped. Its fine if your run Asterisk with a T-1 Card, if not then you are going to experience issues. What Can or Should be Done? To get this solved, Asterisk should obtain its clocking from an internal source in a way that an output stream can be generated without getting any RTP input. The clocking should then be taken from an internal timing mechanism that keeps track of the synchronization. The solution should not require T1 connectivity [IE: no TDM hardware]. Such T1 connectivity would severely limit traffic on the LiveVoip Global SIP network via IP. Developers should work to solve the no alerting scenario's [when peer is set in RCV only mode] and all issues related to the use of silence compression. A configuration option should exist to choose the timing method for customers that want to use Asterisk in calling card applications or any application where no T-1 cards will ever be required. Status: LiveVoip engineers have developed a workaround for our internal switch network. This will be tested and could take up to 14 days to install in every LiveVoip Network Node location. On Tue, 15 Mar 2005 17:07:53 -0500, Robert Webb [EMAIL PROTECTED] wrote: On Tue, 15 Mar 2005 14:50:38 -0700 Daniel Webb [EMAIL PROTECTED] wrote: On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote: Dude, where have you been? This has been discussed here at length. Everyone agrees that it's on LiveVOIP's end, but they're shrugging their shoulders and pointing toward *. Search the list. Could you point out the best way to search the list? Perhaps go to http://lists.digium.com/pipermail/asterisk-users/, go to each month one at a time, then click threads, then do a page search? What a swell interface. How about learning a few Google skills and in the search line type: site:lists.digium.com search criteria THe above site command will only search the url specified. In this case the Asterisk lists. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote: Dude, where have you been? This has been discussed here at length. Everyone agrees that it's on LiveVOIP's end, but they're shrugging their shoulders and pointing toward *. Search the list. Could you point out the best way to search the list? Perhaps go to http://lists.digium.com/pipermail/asterisk-users/, go to each month one at a time, then click threads, then do a page search? What a swell interface. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringback over IAX - LiveVoip
Try googling: QUERY: Asterisk-Users Search String Works quite well. -- www.justfuckinggoogleit.com On Tue, 15 Mar 2005 14:50:38 -0700, Daniel Webb [EMAIL PROTECTED] wrote: On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote: Dude, where have you been? This has been discussed here at length. Everyone agrees that it's on LiveVOIP's end, but they're shrugging their shoulders and pointing toward *. Search the list. Could you point out the best way to search the list? Perhaps go to http://lists.digium.com/pipermail/asterisk-users/, go to each month one at a time, then click threads, then do a page search? What a swell interface. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
On Tue, 15 Mar 2005 14:50:38 -0700 Daniel Webb [EMAIL PROTECTED] wrote: On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote: Dude, where have you been? This has been discussed here at length. Everyone agrees that it's on LiveVOIP's end, but they're shrugging their shoulders and pointing toward *. Search the list. Could you point out the best way to search the list? Perhaps go to http://lists.digium.com/pipermail/asterisk-users/, go to each month one at a time, then click threads, then do a page search? What a swell interface. How about learning a few Google skills and in the search line type: site:lists.digium.com search criteria THe above site command will only search the url specified. In this case the Asterisk lists. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
If asterisk is going to be modified to support LiveVoip expectations, then yet another Dial option would need to be implemented to force ringback to occur as an audio stream for iax only. Guess one could open a bug report for both LiveVoip and Asterisk, but not likely to be addressed any time soon since this is itsp dependent. This would actually be a good idea to to do, simply provide a switch different than 'r' that forces an audio string ringback. Not only would it work for LiveVoip, but for other circumstances as well. Sure, one could argue that we're creating a band-aid for a problem that shouldn't have to exist, but hey, if it works then I don't see why not add it in. People don't need to use the feature if they don't want to... :) FWIW, I submitted a detailed description of the ringback problem to [EMAIL PROTECTED] this morning (March 13). After personally analyzing the ringback issue (and without any feedback from livevoip as yet), my best estimate is that livevoip is going to consider this an asterisk issue as the call is considered answered (due to asterisk's IVR answering the call), and that any further ringing feedback should come from the end nodes (not from the middle- man livevoip switch). That is basically how all real telephony (non- voip providers) would handle such things. That would imply that we'll need to submit a feature request for asterisk to support some sort of inband ringback option selectable by the asterisk implementor. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say are you customer type x?, we don't want you then. Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me. There are more then a few folks using * that try various itsp services without a clue as to how to make things work, and/or with unrealistic expectations. I happen to like their no-nonsense approach on their web site of getting your attention. It sort of resembles some of the postings on this list relative to 'did you try to look for doc'. Overall, I'd give livevoip high marks in both quality and service. Let's hope they can keep it up. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter That is what everyone is bitching about. No matter whether you use the r option or not, you never get ringback through LiveVoIP. And they consistently point the finger at * rather that trying to solve the problem. If you use ethereal to inspect the iax packets in the above case, you see that asterisk is sending an IAX Type=Control packet with a Control subclass: Ringing (3) to the LiveVoip switch. LiveVoip is ignoring that particular control packet. I'd have to guess that LiveVoip wants ringback to occur in the audio stream, not as a iax control packet, and therefore is blaming asterisk. The r option within asterisk (in the above case) is doing exactly what Mark intended for asterisk-to-asterisk iax connections, which is different then LiveVoip expectations. So, who is wrong here, or is this just human translations of what is expected in a non-rfc communications environment? If asterisk is going to be modified to support LiveVoip expectations, then yet another Dial option would need to be implemented to force ringback to occur as an audio stream for iax only. Guess one could open a bug report for both LiveVoip and Asterisk, but not likely to be addressed any time soon since this is itsp dependent. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
Title: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Excellent thing to hear. I am glad there are positives on this site as well as teh warnings. Now to get the ringback issue resolved Using m switch to get MOH is OK but there has to be alogical reason this is occuring adn a way to resolve. Thanks, Wiley From: [EMAIL PROTECTED] on behalf of Rich AdamsonSent: Sat 3/12/2005 5:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say "are you customer type x?, we don't want you then". Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me.There are more then a few folks using * that try various itsp serviceswithout a clue as to how to make things work, and/or with unrealisticexpectations. I happen to like their "no-nonsense" approach on theirweb site of getting your attention. It sort of resembles some of thepostings on this list relative to 'did you try to look for doc'.Overall, I'd give livevoip high marks in both quality and service. Let'shope they can keep it up. :)___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.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 ringback over IAX - LiveVoip
Title: No ringback over IAX - LiveVoip Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
On Fri, 11 Mar 2005 11:47:53 -0700 Wiley Siler [EMAIL PROTECTED] wrote: Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley According to LiveVoip folks it is ASterisk's fault that this does not work. Even though there are many of us using IAX with other providers with no problems. Just don't complain too much as they will email you and tell you that if you do not like it to just cancel. Or, as one user noted here, it seemed that his account was just deleted without any contact. Not sure if that is really what happened or not. But the email contact telling you to cancel will. They did it to me.. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No ringback over IAX - LiveVoip
Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say are you customer type x?, we don't want you then. Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me. Thanks, Wiley -Original Message- From: Robert Webb [mailto:[EMAIL PROTECTED] Sent: Friday, March 11, 2005 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Wiley Siler Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip On Fri, 11 Mar 2005 11:47:53 -0700 Wiley Siler [EMAIL PROTECTED] wrote: Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley According to LiveVoip folks it is ASterisk's fault that this does not work. Even though there are many of us using IAX with other providers with no problems. Just don't complain too much as they will email you and tell you that if you do not like it to just cancel. Or, as one user noted here, it seemed that his account was just deleted without any contact. Not sure if that is really what happened or not. But the email contact telling you to cancel will. They did it to me.. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringback over IAX - LiveVoip
On Fri, 11 Mar 2005, Wiley Siler wrote: I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
I am pretty sure that AAH is adding that automatically. I am checking it out right now. Thank you for the poitner! Chers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Friday, March 11, 2005 1:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip On Fri, 11 Mar 2005, Wiley Siler wrote: I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
On Fri, 11 Mar 2005 21:48:07 +0100 (CET) Peter Svensson [EMAIL PROTECTED] wrote: On Fri, 11 Mar 2005, Wiley Siler wrote: I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter That is what everyone is bitching about. No matter whether you use the r option or not, you never get ringback through LiveVoIP. And they consistently point the finger at * rather that trying to solve the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
Hmm... It could be a simple mistake of course. Is it safe to assume that the host server on their side is Asterisk as well? I assume there are no telco level devices that offer IAX. Could be wrong. If that is the case, it should be easy to infer what is wrong on their side. Obviously someone here has mated a couple of Asterisk boxes before. What would have to be set wrong on their side for the ring tone not to pass? Once that is known, LiveVoIP users can send them the request to fix that particular problem. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Friday, March 11, 2005 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip On Fri, 11 Mar 2005 21:48:07 +0100 (CET) Peter Svensson [EMAIL PROTECTED] wrote: On Fri, 11 Mar 2005, Wiley Siler wrote: I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter That is what everyone is bitching about. No matter whether you use the r option or not, you never get ringback through LiveVoIP. And they consistently point the finger at * rather that trying to solve the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
Maybe the way they present this selection isn't very fortunate, but I fully understand where they're coming from. They're providing wholesale VOIP at wholesale prices. Seriously, do the math -- If you're buying an inbound 800# from them, you could get away with spending only $2.50/month ($29.95 prepaid, expires in a year). And if you used these $2.50 to receive 200 incoming minutes a month, they'll make maybe $1 on you (assuming 1.27c/min sale vs. 0.8c/min cost). If it takes them more than ONE MINUTE/MONTH to support you, they've already blown their profit. I myself have often walked away from expensive customers, and business people much smarter than me do this on a daily basis. -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, March 11, 2005 1:53 PM To: Robert Webb; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say are you customer type x?, we don't want you then. Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me. Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
Using the 'r' option doesn't seem to make a difference for me. To work around this issue, I'm using the 'm' option to play MOH. However it just occurred to me, does anyone have a .mp3 recording of a phone ringing for 30+ seconds? I could play that instead of regular music and it would probably work not too bad. On Fri, 2005-03-11 at 21:48 +0100, Peter Svensson wrote: On Fri, 11 Mar 2005, Wiley Siler wrote: I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] 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] No ringback over IAX - LiveVoip
I agree totally. I just have never seen anyone post it on their site so brazenly. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, March 11, 2005 3:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Maybe the way they present this selection isn't very fortunate, but I fully understand where they're coming from. They're providing wholesale VOIP at wholesale prices. Seriously, do the math -- If you're buying an inbound 800# from them, you could get away with spending only $2.50/month ($29.95 prepaid, expires in a year). And if you used these $2.50 to receive 200 incoming minutes a month, they'll make maybe $1 on you (assuming 1.27c/min sale vs. 0.8c/min cost). If it takes them more than ONE MINUTE/MONTH to support you, they've already blown their profit. I myself have often walked away from expensive customers, and business people much smarter than me do this on a daily basis. -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, March 11, 2005 1:53 PM To: Robert Webb; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say are you customer type x?, we don't want you then. Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me. Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
Dude, where have you been? This has been discussed here at length. Everyone agrees that it's on LiveVOIP's end, but they're shrugging their shoulders and pointing toward *. Search the list. -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, March 11, 2005 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Hmm... It could be a simple mistake of course. Is it safe to assume that the host server on their side is Asterisk as well? I assume there are no telco level devices that offer IAX. Could be wrong. If that is the case, it should be easy to infer what is wrong on their side. Obviously someone here has mated a couple of Asterisk boxes before. What would have to be set wrong on their side for the ring tone not to pass? Once that is known, LiveVoIP users can send them the request to fix that particular problem. Thanks! Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip
I have been on the list and the archive as I stated. Saw the emails but never saw any resolution given as I stated. In case you did not notice, I am suggesting that we can figure out what the problem on their side is and give them the resolution. In that way the problem gets solved and we all benefit. They may honestly believe nothing is wrong. We should focus on solving the problem. Read the other posts before you issue a flame over checking the list. Escpecially since you did not read the list yourself to see my other post or even the fact that I am suggesting we find the problem. I am absolutely fed up with people like you bouncing out these shitty replies like you own the list. If you don't like my question. Delete it. If you don't want to answer. Don't answer. If all you have to say is negative prattling, keep it to yourself. Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, March 11, 2005 3:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Dude, where have you been? This has been discussed here at length. Everyone agrees that it's on LiveVOIP's end, but they're shrugging their shoulders and pointing toward *. Search the list. -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, March 11, 2005 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Hmm... It could be a simple mistake of course. Is it safe to assume that the host server on their side is Asterisk as well? I assume there are no telco level devices that offer IAX. Could be wrong. If that is the case, it should be easy to infer what is wrong on their side. Obviously someone here has mated a couple of Asterisk boxes before. What would have to be set wrong on their side for the ring tone not to pass? Once that is known, LiveVoIP users can send them the request to fix that particular problem. Thanks! Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No ringback over IAX - LiveVoip (or how Wiley made my /dev/null list)
That's very nice and all, but not constructive. Please DO read the posts regarding LiveVOIP's ringback problem. LiveVOIP has even identified as * sending back a non-standard signal. So, we know what the problem is, we know it doesn't exist with other providers, and even in the face of all this information, LiveVOIP seems to be refusing or unable to fix it. That said, there was nothing negative about my post, just stating the very obvious fact that you may have missed part of the discussion. I didn't suggest why you may have missed it, nor did I imply that there's anything wrong with you for missing it. I do, however, take note that you're taking this list much too seriously. If you defend it so vigorously, maybe you could take your arguments off-list rather than plugging up communications with no less than 30 messages in the last 24 hours, the majority of which unconstructive bickering about your personal disagreement with some jerk's response to a newbie. I'm not defending the decision of some to be hostile towards those who are lazy, or even the ridiculously thoughtless sentiment to complain to 10,000 people about some wasted bytes -- but your posts and their frequency are adding a lot of noise to a list which could otherwise be useful. Since you like to give out your recommendations as bullet-points, let me add mine: - If you don't know the answer to a question, don't answer it. - If you have no new information to add, don't post. - If you disagree with someone's attitude or post or whatever, tell that ONE person; don't bother the rest of the free world with your scrupels. - And lastly, don't get offended on this list -- it ain't worth it. Spend more time with your family or pets or car or whatever makes you smile. Have a good weekend. (and yes, you're dev/nul'ed -- it's nothing personal, I'm just trying to cut down on the noise, and at your current rate, you'd add another 1,000 messages/month to this list) -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, March 11, 2005 5:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip I have been on the list and the archive as I stated. Saw the emails but never saw any resolution given as I stated. In case you did not notice, I am suggesting that we can figure out what the problem on their side is and give them the resolution. In that way the problem gets solved and we all benefit. They may honestly believe nothing is wrong. We should focus on solving the problem. Read the other posts before you issue a flame over checking the list. Escpecially since you did not read the list yourself to see my other post or even the fact that I am suggesting we find the problem. I am absolutely fed up with people like you bouncing out these shitty replies like you own the list. If you don't like my question. Delete it. If you don't want to answer. Don't answer. If all you have to say is negative prattling, keep it to yourself. Regards, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] transfer ringback
Not everybody here has voicemail and we have more than one person answering the phones. I would like to know that if Person A transfers to Person B, that after a predefined timeout, it would then transfer the call back to Person A. Right now when a new call comes in, it rings 3 phones. When Person A transfers to Person B, it tries for 15 seconds, then transfers the call back to all 3 people via the Dial command. It's a little confusing to write out, so if more info is needed, I will be happy to elaborate. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Traditional Ringback Tone
I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was generally overlaid on top of real ringing, i.e. 20Hz. So, using the Asterisk example of 420*40, it would seem that a decent ringback would be (420*40)*20. But, of course, that doesn't appear to exist. If it does, I am missing the boat on how to do it properly. So, I have a question: Is it possible to either (a) do the double modulation as listed above, or (b) provide recorded wav or gsm sounds as a background fill while a phone is being rung? I have recordings of various types of older central office ringback tones that I'd just love to be able to put into Asterisk. I know this sounds a bit arcane. But Asterisk can do so many things to order that it really ought to be able to do this, don't you think? Thanks, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.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 ringback on IAX channel after selecting menu option
Here is the call flow: [ivr-incoming] exten = s,1,LookupCIDName exten = s,2,DigitTimeout(2) exten = s,3,ResponseTimeout(10) exten = s,4,Wait(1) exten = s,5,Background(custom/ivr-incoming) exten = 1,1,Background(pls-wait-connect-call) exten = 1,2,Dial(${RINGPHONENUMBERS},20,r) exten = 1,3,Voicemail,u${VMBOX} exten = 1,4,Hangup Running * 1.0.5. The calling party hears the please wait while I connect your call, but does not hear any ringing. I tried inserting exten = 1,1,Ringing but that does not work either. The same call flow from the pstn DOES generate ringback: [fromPSTN] exten = s,1,DigitTimeout(2) exten = s,2,ResponseTimeout(10) exten = s,3,Wait(1) exten = s,4,Background(custom/ivr-greeting) exten = 1,1,Background(pls-wait-connect-call) exten = 1,2,Dial(${RINGPHONENUMBERS},15,r) exten = 1,3,Voicemail,u${VMBOX} exten = 1,4,Hangup Any thoughts. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] disable ringback of held call on zap channel
One Zap FXS channel has dialed to another. Zapata.conf has transfer = yes and threewaycalling = yes. I flash on one of the phones, the other gets the music on hold. If I hang up the flashed phone, it rings back and I am reconnected to the other phone. Is there some way (with flash, not with #) that I could leave the other phone on hold for a longer time? Preferably without having to dial something after the flash. 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] No Ringback tone on Stable 1.0.2
I am noticing that calls that come from our IAX pstn gateway provider and terminate to our Asterisk IVR do not receive ringing when an extension is dialed. For example: An inbound PSTN caller calls our number Asterisk answers and provides greeting PSTN user dials extension of internal SIP phone No ringback is heard from PSTN callers perspective SIP user picks up or the voicemail picks up in the PSTN caller didnt get frustrated and hang up due to silence On the other handIf I call [EMAIL PROTECTED] from an external sip phone all is well as far as ringing goes. I added an r to the dial command to try to force the issue with no success. This is what asterisk says: == CDR updated on IAX2/[EMAIL PROTECTED]/2 -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, SIP/101|20|r) in new stack -- Called 101 -- SIP/101-178a is ringing However, Im still not getting any tone. If you have any other ideas let me know. (I believe the r suggests that it should be sending back tone but my caller is not getting any) BTW Im using the latest stable build 1.0.2 of Asterisk. Thanks, Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.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 ringback problem with Polycom phones and CVS HEAD
For the past week or two, our customers who have Polycom phones have been experiencing a problem... but our customers with Cisco phones do not have this problem. The phones in question are: Polycom SoundPoint IP300 (firmware 1.3.1 or 1.3.4) Polycom SoundPoint IP500 (firmware 1.3.1 or 1.3.4) Cisco 7960 (firmware 7.2 or 7.3) The problem is this: when our Polycom users dial _some_ PSTN numbers, they hear one cycle of ringback, then it's gone. However, the call is still proceeding, and if they wait for it to be answered the call proceeds normally (audio flows in both directions). When they dial _most_ PSTN numbers, this does not happen. In fact, the calls are all following the same path: from the Asterisk server that the phones register to, over IAX to another Asterisk server, then out a PRI (these are all local calls). I have run a sip debug trace of the successful and failing calls, and everything looks normal; there is only one difference for the failing calls. The successful SIP trace looks like this (P-Polycom Phone, A-Asterisk): P-INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 A-SIP/2.0 100 Trying A-SIP/2.0 183 Session Progress A-SIP/2.0 200 OK P-ACK sip:[EMAIL PROTECTED] SIP/2.0 P-BYE sip:[EMAIL PROTECTED] SIP/2.0 A-SIP/2.0 200 OK The failing SIP trace looks like this: P-INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 A-SIP/2.0 100 Trying A-SIP/2.0 183 Session Progress A=SIP/2.0 180 Ringing A-SIP/2.0 200 OK P-ACK sip:[EMAIL PROTECTED] SIP/2.0 P-BYE sip:[EMAIL PROTECTED] SIP/2.0 A-SIP/2.0 200 OK Note the additional 180 Ringing message in this trace. When the Polycom phone receives this, it stops generating (or passing) ringback to the caller. The actual message is this (but my email client has wrapped it): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bK2ec810b2FB407A27;received=68.14.253.125;rport=1172 From: 3011 sip:[EMAIL PROTECTED];tag=B81927F5-872A14B0 To: sip:[EMAIL PROTECTED];user=phone;tag=as4c95cddd Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 This was working fine before a recent upgrade to Asterisk; I believe it started being a problem after I upgraded to CVS HEAD from around 2004-12-10. I assume this difference in the call trace is due to some difference in the call path through the PSTN (one path reports in-band progress, the other out-of-band, or something like that), but I don't understand why the phone would stop ringback when it receives this message. As it stands right now, I'm going to have to suppress these messages completely, as it's not a pleasant problem for my customers to deal with... Anyone have any idea why this message would cause this problem, or what may have changed in chan_sip recently that might have changed the behavior in this area? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Distorted Ringback
Hi, I recently posted to the list about forcing a particular codec along the path of a call and transcoding the voicepath, http://lists.digium.com/pipermail/asterisk-users/2004-October/065218.html, to which I've been able to tame, thanks to the help of the list. I now have a new problem- When I place a call from my grandstream, I hear a distorted ringback. Once the call is answered, everything sounds fine. sip0 an * box which handles registreation and pbx functions. sip1 is the pstn gateway with pri interfaces. Setup is as follows: (iLBC) (ulaw) ++ ++ ++ | gs | - | sip0 | - | sip1 | ++ ++ ++ Nothing is allowed to reinvite in this scenerio, the call path is exactly as you see above. Has anyone run into a problem like this with a similar setup? - Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no ringback - stable not fixed
Several have noted that after updating the Stable version, no ringback tone is heard on sip phones. Adding the ,r option to the Dial() statement corrects the problem. Others have noted upgrading again to the most current stable release also fixes the problem. It does _NOT_ fix the problem as of a few minutes ago. It has been corrected in the development CVS, but not in Stable. FWIW. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ringback
Title: No ringback I just configured Asterisk on a new machine, and other things seem to be working fine, I don't get any audible ringback when dialing calls from a SIP phone or a standard phone connected through a TDM400P. What am I doing wrong here? Thanks -brian
RE: [Asterisk-Users] No ringback
Title: No ringback I had a similar problem. What I did what checked out the version before 03-02-2004. Some change after that date is causing the problem. Gene From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: Saturday, April 03, 2004 4:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No ringback I just configured Asterisk on a new machine, and other things seem to be working fine, I don't get any audible ringback when dialing calls from a SIP phone or a standard phone connected through a TDM400P. What am I doing wrong here? Thanks -brian
RE: [Asterisk-Users] No ringback
Title: No ringback Thanks. Actually,I got the latest from the cvs repository and it's fixed there, too. I suspect that it got broken at some point briefly before someone fixed it. -brian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gene KochanowskySent: Saturday, April 03, 2004 5:25 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] No ringback I had a similar problem. What I did what checked out the version before 03-02-2004. Some change after that date is causing the problem. Gene From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian CuthieSent: Saturday, April 03, 2004 4:32 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] No ringback I just configured Asterisk on a new machine, and other things seem to be working fine, I don't get any audible ringback when dialing calls from a SIP phone or a standard phone connected through a TDM400P. What am I doing wrong here? Thanks -brian
[Asterisk-Users] Missing ringback tone on C7960
Just upgraded to stable CVS-03/20/04-11:54:56 C7960 - * C7960 (all on the same wire), call from on phone to the other does not receive any ringback signal. Total silence while the phone is actually ringing, then hear voicemail anouncements after the 15 second timeout. Was working fine before the upgrade. Anyone else observe the same issue? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Missing ringback tone on C7960
On Mon, Mar 22, 2004 at 01:48:05PM -0600, Rich Adamson wrote: Just upgraded to stable CVS-03/20/04-11:54:56 C7960 - * C7960 (all on the same wire), call from on phone to the other does not receive any ringback signal. Total silence while the phone is actually ringing, then hear voicemail anouncements after the 15 second timeout. Was working fine before the upgrade. Anyone else observe the same issue? Rich I upgraded two servers to the v1-0_stable yesterday. Both servers would not provide ring tone to the callers when I was dialing the 7960 phones. I added the `r` option to the Dial command and that took care of it. Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Missing ringback tone on C7960
On Mon, Mar 22, 2004 at 01:48:05PM -0600, Rich Adamson wrote: Just upgraded to stable CVS-03/20/04-11:54:56 C7960 - * C7960 (all on the same wire), call from on phone to the other does not receive any ringback signal. Total silence while the phone is actually ringing, then hear voicemail anouncements after the 15 second timeout. Was working fine before the upgrade. Anyone else observe the same issue? Rich I upgraded two servers to the v1-0_stable yesterday. Both servers would not provide ring tone to the callers when I was dialing the 7960 phones. I added the `r` option to the Dial command and that took care of it. Thanks Walker... I had to do the same thing. Apparently something changed in the sip area. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Ringback Tone when Dialing (outside caller to internal extension from auto-attendant)
I have been unsuccessful of yet to produce a ringback tone when trying as an outside caller to dial an internal extension from an auto-attendant. This is the scenario: 1. Outside caller dials main line that is going into an FXO card (X100P). 2. Auto-attendant answers 3. Outside caller dials an internal extension 4. Internal extensions are analog phones connected to SPA-2000 using SIP. In Step one, of course the caller gets a ringback tone until the auto-attendant calls (supplied by the PSTN). In Step 3, the extension rings but no ringback tone is supplied to the caller. Things I have tried thus far (_please read_). 1. Removing Answer application from auto-attendant 2. Adding r option to Dial application 3. Add Ringing application before dialing None of these options have worked. Please note that if I call from one internal extension to another, I get a ringback tone. As well if I call an outside line from an internal extension I get a ringback tone. I have searched the Asterisk Wiki, Asterisk Mailing List and google. Any help would be greatly appreciated.
[Asterisk-Users] SIP - Ringback
I am new to the sip side of things and have a question regarding ringback. I don't hear ringback when using the sjphone softphone when dialing internal extensions. It's fine when dialing outside over the pstn. Is this a issue of the softphone, configuration or sip in general? Thank you, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ringback
Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten = s,1,Zapateller(answer|nocallerid) exten = s,2,NoOp exten = s,3,Playback(pls-wait-connect-call) exten = s,4,Dial(${PHONE1}${PHONE2}${PHONE3}${PHONE4},15,Ttm) exten = s,5,Answer exten = s,6,Wait(1) exten = s,7,Voicemail(u${PHONE1VM}) exten = s,8,Hangup exten = s,107,Voicemail(b${PHONE1VM}) exten = s,108,Hangup Do you see anything wrong with this ? Regards...Martin -- Anything is possible on paper. -- Ron McAfee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Ringback on Iconnect or Nikotel
When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback tone when making a call. I have tried adding the r command to the dial sting with no help. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
[Asterisk-Users] No Ringback on Iconnect
When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
[Asterisk-Users] No Ringback on Iconnect
When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
RE: [Asterisk-Users] No Ringback on Iconnect
What is the Exten = .Dial( Line from your extensions.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin Sent: Sunday, October 05, 2003 7:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Ringback on Iconnect When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
RE: [Asterisk-Users] No Ringback on Iconnect
I have tried both of these: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED]||r -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Sunday, October 05, 2003 7:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect What is the Exten = .Dial( Line from your extensions.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Sunday, October 05, 2003 7:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Ringback on Iconnect When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
RE: [Asterisk-Users] No Ringback on Iconnect
Try exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) or exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) It should work -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin Sent: Sunday, October 05, 2003 9:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect I have tried both of these: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED]||r -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Sunday, October 05, 2003 7:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect What is the Exten = .Dial( Line from your extensions.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Sunday, October 05, 2003 7:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Ringback on Iconnect When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
RE: [Asterisk-Users] No Ringback on Iconnect
Changed my conf file to: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED],90,r still no ringback -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Sunday, October 05, 2003 9:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect Try exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) or exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) It should work -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Sunday, October 05, 2003 9:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect I have tried both of these: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED]||r -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Sunday, October 05, 2003 7:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect What is the Exten = .Dial( Line from your extensions.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Sunday, October 05, 2003 7:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Ringback on Iconnect When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
RE: [Asterisk-Users] No Ringback on Iconnect
Then remove the ,r and see if it works. Have you setup your indications.conf? Do you get any messages from asterisk CLI? What device are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin Sent: Sunday, October 05, 2003 11:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect Changed my conf file to: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED],90,r still no ringback -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Sunday, October 05, 2003 9:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect Try exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) or exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) It should work -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Sunday, October 05, 2003 9:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect I have tried both of these: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED]||r -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Sunday, October 05, 2003 7:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringback on Iconnect What is the Exten = .Dial( Line from your extensions.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Sunday, October 05, 2003 7:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Ringback on Iconnect When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no call progress such as ringback when making a call. If I program the SIP phone to directly access iconnect or nikotel I do hear ringing when the outbound call is placed. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
[Asterisk-Users] No Ringback on Iconnect
When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback when making a call. I do see call progress in the console. Does anyone else have this problem or offer any suggestions? Thanks, Kevin
[Asterisk-Users] HELP!!!! Ringback oh323
Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Ringback oh323
Specify option 'r' to dial application. Michael On Friday 01 August 2003 07:13 pm, [EMAIL PROTECTED] wrote: Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] false ringback
Michael Bielicki wrote: Is it possible to give a false ringbakc on asterisk ? What hardware are you using? Petr Michalek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] false ringback
digium cards on * but the carrier connections are all h323 and some of them don't provide ringback. On Thursday 03 Apr 2003 21:42, Petr Michálek shaped the electrons to say: Michael Bielicki wrote: Is it possible to give a false ringbakc on asterisk ? What hardware are you using? Petr Michalek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users