Hello,
While trying to install H323 support for asterisk, I missed one step. After compiling files in channel/h323, need to select chanh323 from the menu and compile and install asterisk. cd asterisk * ./configure * make menuconfig channel drivers->chanh323 save the setting by giving x. * make make * make install Hope this will help someone. Thanks & Regards Bincy K Philip -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, November 13, 2007 2:56 PM To: [email protected] Subject: asterisk-users Digest, Vol 40, Issue 34 Send asterisk-users mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: asterisk and installing chan_h323.so rpm (Dovid B) 2. Re: Asterisk direct dialing (Dovid B) 3. Re: Help: Static and dropped calls (Dovid B) 4. Re: How to delete voice mail messages? (Dovid B) 5. Re: asterisk and installing chan_h323.so rpm (David Boyd) 6. Re: ztdummy and BackGround (Tony Plack) 7. MOH Codec Issue (Nick Brown) 8. Re: RTP traffic not being forwarded (Ryan Newington) 9. Re: MOH Codec Issue (Paul Hales) 10. Chatterbug (Robert Goodyear) 11. Re: MOH Codec Issue (Nick Brown) 12. Re: Chatterbug (Paul Hales) 13. Re: MOH Codec Issue (Paul Hales) 14. Re: MOH Codec Issue (Paul Hales) 15. Stress-Testing Asterisk (Jeng Yu) 16. Re: Stress-Testing Asterisk (Suity Zsolt) 17. Re: Stress-Testing Asterisk (Tzafrir Cohen) 18. Toshiba DK - Asterisk Integration (Indika Wasala) 19. Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11 (Erik Wartusch) ---------------------------------------------------------------------- Message: 1 Date: Tue, 13 Nov 2007 03:00:22 +0200 From: "Dovid B" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] asterisk and installing chan_h323.so rpm To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original ----- Original Message ----- From: "Bincy K. Philip" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Thursday, November 08, 2007 2:13 PM Subject: [asterisk-users] asterisk and installing chan_h323.so rpm > Hello, > > When I tried to install chan_h323-1.0.1-module.i386 RPM i got these > errors. > > > Failed dependencies: > libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 > libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 > > But i found the same files in > > /usr/lib/libh323_linux_x86_r.so.1 > /usr/lib/libpt_linux_x86_r.so.1 > > > What to do for asterisk to detect the same files? > I had this issue in the past. I do not remember what I did to resolve it. In the end I went with the h323 channel driver located in the asterisk add-ons. It was a lot easier to work with and worked with out any issues. ------------------------------ Message: 2 Date: Tue, 13 Nov 2007 03:02:08 +0200 From: "Dovid B" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Asterisk direct dialing To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original ----- Original Message ----- From: "Gopal krishnan" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Saturday, November 10, 2007 12:46 PM Subject: [asterisk-users] Asterisk direct dialing > Hi, > I am using Asterisk 1.2.24, I have written my dialplan to land > with an IVR with the same time if the customer knows the parties > extensions they can dial directly, but what happens is sometimes its > working and sometime its not working. > My extensions.conf as follows, > > [incoming] > exten => 052477302,1,Wait(2) > exten => 052477302,2,NoOp(${CALLERIDNUM}) > exten => 052477302,3,Goto(from-internal,s,1) > > [from-internal] > exten => s,1,Answer() > exten => s,2,Background(welcome_pride5) > exten => 501,1,Dial(Zap/1) > exten => 502,1,Dial(Zap/2) > exten => 503,1,Dial(Zap/3) > exten => 504,1,Dial(Zap/4) > exten => 505,1,Dial(Zap/5) > > I dont know what could be the reason. Is there any other way that i can > use. > > -- > Thank you, with regards, > Gopal, > What comes up in the CLI when it does not work ? ------------------------------ Message: 3 Date: Tue, 13 Nov 2007 03:10:36 +0200 From: "Dovid B" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Help: Static and dropped calls To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" You need to provide more information that just that. Maybe a CLI output ? Have you tested with any other providers ? SIP debug ? Ran a trace ? We aren't mind readers here. ----- Original Message ----- From: Jarga Jallow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 05, 2007 8:27 PM Subject: [asterisk-users] Help: Static and dropped calls Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow ------------------------------------------------------------------------------ _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/3db40dc2/attachment-0001.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 3781 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/3db40dc2/attachment-0001.jpeg ------------------------------ Message: 4 Date: Tue, 13 Nov 2007 03:14:48 +0200 From: "Dovid B" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] How to delete voice mail messages? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" I wrote this for a client a while back: [del-all-vm] exten => s,1,Set(TIMEOUT(digit)=3) exten => s,2,Set(TIMEOUT(response)=6) exten => s,3,Background(enter-exten-for-vm-to-delete) exten => _XX,1,Set(THIER_EXTEN=${EXTEN}) exten => _XX,2,Goto(del-all-vm-confirm,s,1) exten => i,1,Playback(invalid) exten => i,2,Goto(s,1) exten => t,1,Goto(s,1) [del-all-vm-confirm] exten => s,1,Set(TIMEOUT(digit)=3) exten => s,2,Set(TIMEOUT(response)=6) exten => s,3,Background(are-you-sure-del-all-vm) exten => s,4,Saynumber(${THIER_EXTEN}) exten => s,5,Background(1-for-yes-2-for-no) exten => 1,1,System(rm -rf /var/spool/asterisk/default/techmast/${THIER_EXTEN}/INBOX/*.*) exten => 1,2,Playback(all-vm-deleted) exten => 1,3,Congestion exten => 1,4,Hangup exten => 2,1,Playback(close-call-not-deleted) exten => 2,2,Congestion exten => 2,3,Hangup exten => i,1,Playback(invalid) exten => i,2,Goto(s,1) exten => t,1,Goto(s,1) ----- Original Message ----- From: voip crazy To: [email protected] Sent: Monday, November 05, 2007 1:15 PM Subject: [asterisk-users] How to delete voice mail messages? Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any "messages index file" or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. ------------------------------------------------------------------------------ _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/ff26de0c/attachment-0001.htm ------------------------------ Message: 5 Date: Mon, 12 Nov 2007 20:30:25 -0500 (EST) From: "David Boyd" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] asterisk and installing chan_h323.so rpm To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain;charset=iso-8859-1 > > ----- Original Message ----- > From: "Bincy K. Philip" <[EMAIL PROTECTED]> > To: <[email protected]> > Sent: Thursday, November 08, 2007 2:13 PM > Subject: [asterisk-users] asterisk and installing chan_h323.so rpm > > >> Hello, >> >> When I tried to install chan_h323-1.0.1-module.i386 RPM i got these >> errors. >> >> >> Failed dependencies: >> libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 >> libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 >> >> But i found the same files in >> >> /usr/lib/libh323_linux_x86_r.so.1 >> /usr/lib/libpt_linux_x86_r.so.1 >> >> >> What to do for asterisk to detect the same files? >> > I had this issue in the past. I do not remember what I did to resolve it. > In > the end I went with the h323 channel driver located in the asterisk > add-ons. > It was a lot easier to work with and worked with out any issues. > It seems to me that you need to run ldconfig so as to pick up the location of the specified libraries. Do a google on it to see syntax of "man ldconfig". You could also hack things by linking to the libraries from the expected directories (What the rpm is expecting) if executing ldconfig doesn't work. db > ------------------------------ Message: 6 Date: Mon, 12 Nov 2007 20:25:49 -0600 From: Tony Plack <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] ztdummy and BackGround To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071112/2ef88eab/attachment-0001.htm ------------------------------ Message: 7 Date: Tue, 13 Nov 2007 15:04:56 +1100 From: Nick Brown <[EMAIL PROTECTED]> Subject: [asterisk-users] MOH Codec Issue To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/6dfe45b7/attachment-0001.htm ------------------------------ Message: 8 Date: Tue, 13 Nov 2007 15:26:52 +1100 From: Ryan Newington <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] RTP traffic not being forwarded To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Hi Vivek, Thanks for the link. I had a look through and couldn't find anything that worked. There are no NAT problems as this is all taking place on my internal network. The rtp.conf is used to configure the ports. There are no firewalls or gateways in between these devices. Asterisk is listening on the correct ports, and receiving the traffic, as no ICMP messages are being generated to say that the packets could not be delivered. Ryan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava Sent: Monday, 12 November 2007 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded well i think rtp port range is defined in rtp.conf and correct me if i am wrong, these ports must be opened/forwarded to communicate. http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Let me know if you need more information. Thanks, Vivek On 11/11/07, Ryan Newington <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: Hi Vivek, I'm not sure what you mean, could you explain further? Regards Ryan From: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>] On Behalf Of Vivek Shrivastava Sent: Monday, 12 November 2007 1:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf) Regards, Vivek On 11/11/07, Ryan Newington <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: Hi Vivek, The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000. I can change the port range on the media server, asterisk and the device, but neither seems to help. My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk) SIP Phone <---> Media Gateway ---> Asterisk <--- SIP Phone Ryan From: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>] On Behalf Of Vivek Shrivastava Sent: Sunday, 11 November 2007 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help. --Vivek On 11/10/07, Ryan Newington <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: Hi Luki, Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success. Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway. SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone An asterisk internal call will work fine. Eg; SIP Phone <-> Asterisk <-> SIP Phone Regards Ryan -----Original Message----- From: [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> [mailto: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ] On Behalf Of Luki Sent: Sunday, 11 November 2007 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded > When using 'rtp debug' on the asterisk console, it shows that it is > receiving traffic from one endpoint, but not the other. A wireshark trace > reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's "on the wire", i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com--<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-- <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-- <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--<http://www.api-digital.com--/> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/f3790661/attachment-0001.htm ------------------------------ Message: 9 Date: Tue, 13 Nov 2007 15:56:21 +1100 From: Paul Hales <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] MOH Codec Issue To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Cc: Nick Brown <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain What format is your music on hold in? PaulH On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: > Afternoon All, > > Today rolled a pre-production box from Trunk back to 1.4.7 (In an > attempt to get a working SCCP channel). During the process Music On > Hold appears to have died (Not, just when calling from a SCCP device, > but coming in on SIP also). > > CLI is showing > > -- Executing [EMAIL PROTECTED]:2] > MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack > [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to > find a codec translation path from alaw to unknown > [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: > Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' > -- Started music on hold, class '?S?', on channel > 'SIP/10.97.1.33-09f0cfc8' > [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: > Unable to start music on hold (class 'sounds') on channel > SIP/10.97.1.33-09f0cfc8 > > Have attempted to use an alternate Music On Hold context and forced a > format= within musiconhold.conf. > > Otherwise all other audio (Playback, voice etc) seems fine. > > Anyone seen this before? Can not see anything in the tracker regarding > this issue in 1.4.7 specifically. > > Cheers > Nick. > _______________________________________________ > --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 ------------------------------ Message: 10 Date: Mon, 12 Nov 2007 21:07:11 -0800 From: Robert Goodyear <[EMAIL PROTECTED]> Subject: [asterisk-users] Chatterbug To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Does anyone know anything about the Chatterbug product? I can't tell if it's an ATA with a modem or some sort of LCR proxy or somesuch. Anyone? ------------------------------ Message: 11 Date: Tue, 13 Nov 2007 16:20:42 +1100 From: Nick Brown <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] MOH Codec Issue To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="ISO-8859-1" It was using the 3 wav's from Freeplay. I have just recompiled and told it to pull down the ULAW versions, then removed the Wav's however it has made no difference. Cheers Nick On 13/11/07 3:56 PM, "Paul Hales" wrote: > > What format is your music on hold in? > > PaulH > > > On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: >> Afternoon All, >> >> Today rolled a pre-production box from Trunk back to 1.4.7 (In an >> attempt to get a working SCCP channel). During the process Music On >> Hold appears to have died (Not, just when calling from a SCCP device, >> but coming in on SIP also). >> >> CLI is showing >> >> -- Executing [EMAIL PROTECTED]:2] >> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack >> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to >> find a codec translation path from alaw to unknown >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: >> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' >> -- Started music on hold, class '?S?', on channel >> 'SIP/10.97.1.33-09f0cfc8' >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: >> Unable to start music on hold (class 'sounds') on channel >> SIP/10.97.1.33-09f0cfc8 >> >> Have attempted to use an alternate Music On Hold context and forced a >> format= within musiconhold.conf. >> >> Otherwise all other audio (Playback, voice etc) seems fine. >> >> Anyone seen this before? Can not see anything in the tracker regarding >> this issue in 1.4.7 specifically. >> >> Cheers >> Nick. >> _______________________________________________ >> --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 > Regards, Nick Brown Ipera Communications Pty Ltd Level 1, 9 Denison Street,????????????? Newcastle West NSW 2302 PO Box 2115, Dangar NSW 2309??????? ? P: +61 2 4910 1000 ? F: +61 2 4910 1099 ? ABN: 31 090 964 104 ------------------------------ Message: 12 Date: Tue, 13 Nov 2007 16:21:07 +1100 From: Paul Hales <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Chatterbug To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf PaulH On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote: > Does anyone know anything about the Chatterbug product? I can't tell > if it's an ATA with a modem or some sort of LCR proxy or somesuch. > > Anyone? > > > _______________________________________________ > --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 ------------------------------ Message: 13 Date: Tue, 13 Nov 2007 16:48:58 +1100 From: Paul Hales <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] MOH Codec Issue To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=utf-8 Is it possibly a funny zaptel issue? Paul Hales AsteriskIT On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote: > It was using the 3 wav's from Freeplay. I have just recompiled and told it > to pull down the ULAW versions, then removed the Wav's however it has made > no difference. > > Cheers > Nick > > On 13/11/07 3:56 PM, "Paul Hales" wrote: > > > > > What format is your music on hold in? > > > > PaulH > > > > > > On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: > >> Afternoon All, > >> > >> Today rolled a pre-production box from Trunk back to 1.4.7 (In an > >> attempt to get a working SCCP channel). During the process Music On > >> Hold appears to have died (Not, just when calling from a SCCP device, > >> but coming in on SIP also). > >> > >> CLI is showing > >> > >> -- Executing [EMAIL PROTECTED]:2] > >> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack > >> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to > >> find a codec translation path from alaw to unknown > >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: > >> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' > >> -- Started music on hold, class '?S?', on channel > >> 'SIP/10.97.1.33-09f0cfc8' > >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: > >> Unable to start music on hold (class 'sounds') on channel > >> SIP/10.97.1.33-09f0cfc8 > >> > >> Have attempted to use an alternate Music On Hold context and forced a > >> format= within musiconhold.conf. > >> > >> Otherwise all other audio (Playback, voice etc) seems fine. > >> > >> Anyone seen this before? Can not see anything in the tracker regarding > >> this issue in 1.4.7 specifically. > >> > >> Cheers > >> Nick. > >> _______________________________________________ > >> --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 > > > > Regards, > Nick Brown > > Ipera Communications Pty Ltd > Level 1, 9 Denison Street, > Newcastle West NSW 2302 > PO Box 2115, Dangar NSW 2309 > > ? P: +61 2 4910 1000 > ? F: +61 2 4910 1099 > ? ABN: 31 090 964 104 > > > _______________________________________________ > --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 > ------------------------------ Message: 14 Date: Tue, 13 Nov 2007 16:48:58 +1100 From: Paul Hales <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] MOH Codec Issue To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=utf-8 Is it possibly a funny zaptel issue? Paul Hales AsteriskIT On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote: > It was using the 3 wav's from Freeplay. I have just recompiled and told it > to pull down the ULAW versions, then removed the Wav's however it has made > no difference. > > Cheers > Nick > > On 13/11/07 3:56 PM, "Paul Hales" wrote: > > > > > What format is your music on hold in? > > > > PaulH > > > > > > On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: > >> Afternoon All, > >> > >> Today rolled a pre-production box from Trunk back to 1.4.7 (In an > >> attempt to get a working SCCP channel). During the process Music On > >> Hold appears to have died (Not, just when calling from a SCCP device, > >> but coming in on SIP also). > >> > >> CLI is showing > >> > >> -- Executing [EMAIL PROTECTED]:2] > >> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack > >> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to > >> find a codec translation path from alaw to unknown > >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: > >> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' > >> -- Started music on hold, class '?S?', on channel > >> 'SIP/10.97.1.33-09f0cfc8' > >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: > >> Unable to start music on hold (class 'sounds') on channel > >> SIP/10.97.1.33-09f0cfc8 > >> > >> Have attempted to use an alternate Music On Hold context and forced a > >> format= within musiconhold.conf. > >> > >> Otherwise all other audio (Playback, voice etc) seems fine. > >> > >> Anyone seen this before? Can not see anything in the tracker regarding > >> this issue in 1.4.7 specifically. > >> > >> Cheers > >> Nick. > >> _______________________________________________ > >> --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 > > > > Regards, > Nick Brown > > Ipera Communications Pty Ltd > Level 1, 9 Denison Street, > Newcastle West NSW 2302 > PO Box 2115, Dangar NSW 2309 > > ? P: +61 2 4910 1000 > ? F: +61 2 4910 1099 > ? ABN: 31 090 964 104 > > > _______________________________________________ > --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 > ------------------------------ Message: 15 Date: Tue, 13 Nov 2007 07:22:17 +0000 (GMT) From: Jeng Yu <[EMAIL PROTECTED]> Subject: [asterisk-users] Stress-Testing Asterisk To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=iso-8859-1 Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Thanks, Jeng ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ------------------------------ Message: 16 Date: Tue, 13 Nov 2007 09:00:06 +0100 From: Suity Zsolt <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Stress-Testing Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Jeng Yu wrote: > Hi All, > > I was wondering, what tools are readily available out > there in Asteriskland for me to use in stress/load > testing asterisk box I have in the lab. I want to > observe how my box holds out under heavy/light/medium > load. Try SIPp from HP (http://sipp.sourceforge.net/index.html) or SIP swiss army knife - SIPSAK (http://sipsak.org/) -- Suich ------------------------------ Message: 17 Date: Tue, 13 Nov 2007 10:21:17 +0200 From: Tzafrir Cohen <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Stress-Testing Asterisk To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Tue, Nov 13, 2007 at 09:00:06AM +0100, Suity Zsolt wrote: > Jeng Yu wrote: > > Hi All, > > > > I was wondering, what tools are readily available out > > there in Asteriskland for me to use in stress/load > > testing asterisk box I have in the lab. I want to > > observe how my box holds out under heavy/light/medium > > load. > > Try SIPp from HP (http://sipp.sourceforge.net/index.html) > or SIP swiss army knife - SIPSAK (http://sipsak.org/) and also: Asterisk (http://asterisk.org/) is a highly programmable and configurable PBX. Use one on a stornger box or several "client" boxes. This can allow you to stress a box through SIP, IAX, H323, or whatever. (/me wonders if NBS could be used to stress-test an Asterisk box :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ------------------------------ Message: 18 Date: Tue, 13 Nov 2007 13:56:02 +0530 From: Indika Wasala <[EMAIL PROTECTED]> Subject: [asterisk-users] Toshiba DK - Asterisk Integration To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the Digium cards I should use in each of these cases and How should I integrate Asterisk with above systems. I read the article in http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata and not sure whether that scenario fits mine. Also it is bit confusing to identify what Digium cards should I need for my cases. Any help is highly appreciated. Thanks, Indika. ------------------------------ Message: 19 Date: Tue, 13 Nov 2007 10:25:33 +0100 From: Erik Wartusch <[EMAIL PROTECTED]> Subject: [asterisk-users] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11 To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest "official" release you can expect a few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped calls). So you have to install the BETAS whether you want or not... That you have to use unique ports is a rumour and not SIP standard. As John said --> IP:Port must be unique . I definitely not understand why I should use random ports. Kind Regards, Erik ____________ > I`m using several GXP2020 phones with newest Firmware 1.1.4.18. I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22 and have eliminated that. > Asterisk Version: 1.4.11. Me too. Still testing 1.4.13 on a non-production system. > I use on every phone the 10000 as local port and in the rtp.conf > >From my knowledge of IP I don't think this is a problem since the address/port would be unique. However the example config I originally had from Grandstream indicated that each phone should use a different port and recommended to use the random port option on the phones. I have since assigned the port number on each phone to 10000 plus the extension number. This was done to create a unique port number and to help with troubleshooting when using Wireshark or tcpdump. I set this in the config file for each phone. John _______________________________________________ --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 End of asterisk-users Digest, Vol 40, Issue 34 ********************************************** _______________________________________________ --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
