Re: [asterisk-users] html/js/flash/air SIP clients?
I am also interested in this. If you don't mind, would you be able to share your findings? Thanks, Kannan. On Thu, Aug 2, 2012 at 7:57 AM, Arstan Jusupov arst...@gmail.com wrote: Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -- _ -- 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] html/js/flash/air SIP clients?
On Thu, 2 Aug 2012 10:27:59 +0800 Arstan Jusupov arst...@gmail.com wrote: Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. http://www.sipml5.org/ -- _ -- 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] can't get libpri/PRI to work
Hi, I am using Red Hat Enterprise Linux 5.5 (32bit) and Asterisk 1.8.12, Dahdi 2.4.1.2 and libpri 1.4.2. The installations is fine. But in the Asterisk CLI prompt the pri commands are missing, only the pri intense debug span is populated. Even if i execute that command it results me to pri set debug 2 span 1 is not a valid command Any assistance would be appreciated. Regards, Gopal. -- _ -- 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] DTMF transmission problem
I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN -- Metaswitch -SIP- Asterisk -SIP or IAX2- CPE Asterisk is running Asterisk 10.4.0 on a CentOS 6.2 VM residing on a CentOS 6.3 KVM host. Asterisk has one network interface connected to the Metaswitch without NAT to place/receive calls from the PSTN, and a separate interface to connect to CPE equipment. SIP and IAX are bound to both interfaces. Vocal call quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the PSTN to the customer isn't. Asterisk is set to remain in the media path on all calls. The customer facing IP address on the Asterisk server is private and is being 1:1 NATed through a MikroTik RB 1100 to a public address that the customers are then connecting to. I have also placed test calls with the customer equipment inside the same LAN as the Asterisk server's customer facing IP address (no NAT) with precisely the same symptoms. The same symptoms persist whether the PSTN or the CPE initiate the call. My example configs are as follows: SIP - [general] limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes disallow=all allow=g722 allow=ulaw allow=gsm allowoverlap=no callevents=yes allowguest=no directmedia=no bindport=bind_here bindaddr=to_this_address srvlookup=yes maxexpiry=7200 defaultexpiry=3600 [authentication] [test-voice] type=friend host=dynamic secret=not_my_secret context=users disallow=all allow=ulaw nat=yes directmedia=no qualify=yes trunk=no IAX2 - [general] bindport=bind_here bindaddr=to_this_address delayreject=yes disallow=all allow=g722 allow=ulaw allow=gsm jitterbuffer=no encryption=yes [test-fax1] type=friend host=dynamic username=test-fax1 secret=not_my_secret context=users disallow=all allow=ulaw qualify=yes trunk=no requirecalltoken=no SIP peers are Zhone ZNID-2xxx series ONTs. IAX peers are ATCOM AG198 ATA gateways, either behind the ONTs (but on the same voice VLAN the ONTs use to talk to Asterisk) or on my Asterisk server's local network. The voice VLAN is a different subnet than Asterisk is on, but no NAT exists between the subnets. Thank you, Noah Engelberth System Administration MetaLINK Technologies -- _ -- 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] can't get libpri/PRI to work
Finally we made it work by enabling chan_dahdi.so file in asterisk make menuselect. But the confusion what I have is without doing this, when I unload and load chan_dahdi.so file I didn'd faced any error. Anyways now it is working. Regards, Gopal. On Thu, Aug 2, 2012 at 5:58 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am using Red Hat Enterprise Linux 5.5 (32bit) and Asterisk 1.8.12, Dahdi 2.4.1.2 and libpri 1.4.2. The installations is fine. But in the Asterisk CLI prompt the pri commands are missing, only the pri intense debug span is populated. Even if i execute that command it results me to pri set debug 2 span 1 is not a valid command Any assistance would be appreciated. Regards, Gopal. -- _ -- 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] DTMF transmission problem
On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote: I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN -- Metaswitch -SIP- Asterisk -SIP or IAX2- CPE [snip] ... Vocal call quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the PSTN to the customer isn't ... [snip] The same symptoms persist whether the PSTN or the CPE initiate the call. What is the dtmf mode of Metaswitch in the above diagram? Is it possible that it's muting the DTMF and then not generating the corresponding DTMF event messages? Everytime I've seen clipped DTMF in the past it was due to imperfect muting at the PSTN - SIP interface. You should be able to take a packet trace on the interface of the Asterisk server communicating with the Metaswitch to determine whether the problem first appears at the switch or in your Asterisk server. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] DTMF transmission problem
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, August 02, 2012 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF transmission problem On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote: I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN -- Metaswitch -SIP- Asterisk -SIP or IAX2- CPE [snip] ... Vocal call quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the PSTN to the customer isn't ... [snip] The same symptoms persist whether the PSTN or the CPE initiate the call. What is the dtmf mode of Metaswitch in the above diagram? Is it possible that it's muting the DTMF and then not generating the corresponding DTMF event messages? Everytime I've seen clipped DTMF in the past it was due to imperfect muting at the PSTN - SIP interface. According to the gentleman that manages the Metaswitch, it's set to allow for either in or out of band dtmf. Based on the packet trace, the packets are coming across as RFC 2833 RTP events. Aside from the very first digit, which Wireshark shows as 7 RTP Event packets and 3 RTP Event (end) packets, all the other ones on my test call came across as 8 RTP Event packets and 3 RTP Event (end) packets. All of the RTP Event packets are in sequence for the call's RTP stream. Also, when I'm monitoring in Asterisk, if I configure logger.conf to output DTMF events into the console, Asterisk is recognizing the DTMF: [Aug 2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4' received on SIP/PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4146 __ast_read: DTMF begin passthrough '4' on SIP/ PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4051 __ast_read: DTMF end '4' received on SIP/ PSTN-SIP-PEER, duration 280 ms [Aug 2 12:25:25] DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted with begin '4' on SIP/ PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4120 __ast_read: DTMF end passthrough '4' on SIP/ PSTN-SIP-PEER You should be able to take a packet trace on the interface of the Asterisk server communicating with the Metaswitch to determine whether the problem first appears at the switch or in your Asterisk server. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- __ ___ -- 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] DTMF transmission problem
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Noah Engelberth Sent: Thursday, August 02, 2012 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF transmission problem -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, August 02, 2012 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF transmission problem On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote: I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN -- Metaswitch -SIP- Asterisk -SIP or IAX2- CPE [snip] ... Vocal call quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the PSTN to the customer isn't ... [snip] The same symptoms persist whether the PSTN or the CPE initiate the call. What is the dtmf mode of Metaswitch in the above diagram? Is it possible that it's muting the DTMF and then not generating the corresponding DTMF event messages? Everytime I've seen clipped DTMF in the past it was due to imperfect muting at the PSTN - SIP interface. According to the gentleman that manages the Metaswitch, it's set to allow for either in or out of band dtmf. Based on the packet trace, the packets are coming across as RFC 2833 RTP events. Aside from the very first digit, which Wireshark shows as 7 RTP Event packets and 3 RTP Event (end) packets, all the other ones on my test call came across as 8 RTP Event packets and 3 RTP Event (end) packets. All of the RTP Event packets are in sequence for the call's RTP stream. Also, when I'm monitoring in Asterisk, if I configure logger.conf to output DTMF events into the console, Asterisk is recognizing the DTMF: [Aug 2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4' received on SIP/PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4146 __ast_read: DTMF begin passthrough '4' on SIP/ PSTN-SIP- PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4051 __ast_read: DTMF end '4' received on SIP/ PSTN-SIP-PEER, duration 280 ms [Aug 2 12:25:25] DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted with begin '4' on SIP/ PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4120 __ast_read: DTMF end passthrough '4' on SIP/ PSTN-SIP-PEER Additional information I discovered after my previous reply: I have a separate Asterisk VM instance (in all other ways the same implementation as above) that is running an IVR. This instance has no issues with inbound DTMF within the IVR, but does exhibit the same symptoms for DTMF when bridged through to an IAX2 peer with the same settings as the first Asterisk VM. On the second Asterisk (with the IVR), DTMF to my Cisco/Linksys SPA942 SIP phones works properly, but not to the IAX or SIP ATAs that I am using (the same ones I'm having problems with on the first Asterisk). All of the live customers on the first Asterisk are ATAs, so I don't know as of this time whether or not SPA phones are working correctly on the first server, though it's reasonable to assume they are. In addition, calls from an SPA942 phone to the IAX or SIP ATAs are also not transmitting DTMF to the ATA device's endpoint. DTMF from the ATA device's endpoint to the SPA942 is working correctly, as is both directions of voice audio. You should be able to take a packet trace on the interface of the Asterisk server communicating with the Metaswitch to determine whether the problem first appears at the switch or in your Asterisk server. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- -- _ -- 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] Originate call from cli does not work for SIP line...
I have a SIP line that is working fine when I make calls from IP phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error: originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received response: Forbidden from 'Anonymous sip:XX@anonymous.invalid;tag=as79fffc8d' Here is the sip.conf entry for that line: [protel-out] defaultuser=XX secret= fromuser=XX type=peer fromdomain=i2next.com.mx host=i2next.com.mx disallowed_methods = UPDATE nat=no qualify=no insecure=port,invite directmedia=no disallow=all allow=g729 context=entrada trustrpid=yes sendrpid=yes As I mentioned it works if I dial from a phone. CLI or AMI fails. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] html/js/flash/air SIP clients?
That is great! Thank you. On Thu, Aug 2, 2012 at 12:19 PM, Lefteris Zafiris zaf@gmail.com wrote: On Thu, 2 Aug 2012 10:27:59 +0800 Arstan Jusupov arst...@gmail.com wrote: Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. http://www.sipml5.org/ -- _ -- 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] DTMF transmission problem
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Noah Engelberth Sent: Thursday, August 02, 2012 1:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF transmission problem -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Noah Engelberth Sent: Thursday, August 02, 2012 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF transmission problem -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, August 02, 2012 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF transmission problem On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote: I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN -- Metaswitch -SIP- Asterisk -SIP or IAX2- CPE [snip] ... Vocal call quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the PSTN to the customer isn't ... [snip] The same symptoms persist whether the PSTN or the CPE initiate the call. What is the dtmf mode of Metaswitch in the above diagram? Is it possible that it's muting the DTMF and then not generating the corresponding DTMF event messages? Everytime I've seen clipped DTMF in the past it was due to imperfect muting at the PSTN - SIP interface. According to the gentleman that manages the Metaswitch, it's set to allow for either in or out of band dtmf. Based on the packet trace, the packets are coming across as RFC 2833 RTP events. Aside from the very first digit, which Wireshark shows as 7 RTP Event packets and 3 RTP Event (end) packets, all the other ones on my test call came across as 8 RTP Event packets and 3 RTP Event (end) packets. All of the RTP Event packets are in sequence for the call's RTP stream. Also, when I'm monitoring in Asterisk, if I configure logger.conf to output DTMF events into the console, Asterisk is recognizing the DTMF: [Aug 2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4' received on SIP/PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4146 __ast_read: DTMF begin passthrough '4' on SIP/ PSTN-SIP- PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4051 __ast_read: DTMF end '4' received on SIP/ PSTN-SIP-PEER, duration 280 ms [Aug 2 12:25:25] DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted with begin '4' on SIP/ PSTN-SIP-PEER [Aug 2 12:25:25] DTMF[19319]: channel.c:4120 __ast_read: DTMF end passthrough '4' on SIP/ PSTN-SIP-PEER Additional information I discovered after my previous reply: I have a separate Asterisk VM instance (in all other ways the same implementation as above) that is running an IVR. This instance has no issues with inbound DTMF within the IVR, but does exhibit the same symptoms for DTMF when bridged through to an IAX2 peer with the same settings as the first Asterisk VM. On the second Asterisk (with the IVR), DTMF to my Cisco/Linksys SPA942 SIP phones works properly, but not to the IAX or SIP ATAs that I am using (the same ones I'm having problems with on the first Asterisk). All of the live customers on the first Asterisk are ATAs, so I don't know as of this time whether or not SPA phones are working correctly on the first server, though it's reasonable to assume they are. In addition, calls from an SPA942 phone to the IAX or SIP ATAs are also not transmitting DTMF to the ATA device's endpoint. DTMF from the ATA device's endpoint to the SPA942 is working correctly, as is both directions of voice audio. You should be able to take a packet trace on the interface of the Asterisk server communicating with the Metaswitch to determine whether the problem first appears at the switch or in your Asterisk server. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- -- At the risk of answering myself -- Found that on calls Asterisk was bridging together and not hearing the DTMF, it was working normally. On calls that Asterisk was still hearing the DTMF, it was being clipped. It seems that Asterisk was involved in relaying the DTMF on calls to IAX endpoints as well as calls to SIP endpoints on a different network from the Asterisk
Re: [asterisk-users] No audio playing back voicemail from odbc
Well, it gets even stranger I've installed version 10.2.1, instead of 10.7.1, and copied the configuration files from another identical server that is running 10.2.1 is working correctly. I STILL can't get voicemail to play back. I can hear the password prompts Theses are, what I think to be, the relevant settings in voicemail.conf: ;minmessage=3 maxsilence=10 silencethreshold=128 When I set silencethresholdo either 500 or 64, I still didn't hear anything. (But I did hear several seconds of actual silence. The .wav file contains nothing but silence. So, fiddling with the silencethreshold in both directions, doesn't seem to change the symptoms. Where else should I look? TIA Mike. On Saturday 28 July 2012 3:43:55 am Support wrote: Hi all, I'm trying to get my voicemail messages stored in a mysql database via odbc. Most of it is working, except, when I play my voicemail messages, I don't hear anything. I can confirm that the messages are getting stored in the database: select uniqueid,msgnum,dir,context,macrocontext,callerid,origtime,duration,mailbox user,mailboxcontext,flag,`read` from voicemessages; +--++-+ ---+--+++--+--- --++--+--+ | uniqueid | msgnum | dir | context | macrocontext | callerid | origtime | duration | mailboxuser | mailboxcontext | flag | read | +--++-+ ---+--+++--+--- --++--+--+ |7 | 0 | /var/spool/asterisk/voicemail/diehlnet/7001/Old | customers | | 17442025-1 | 1343462922 | 0| 7001 | diehlnet | |0 | +--++-+ ---+--+++--+--- --++--+--+ select length(recording) from voicemessages; +---+ | length(recording) | +---+ | 52204 | +---+ This is what the console displays during message playback: [Jul 28 02:28:35] == Parsing '/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.txt': [Jul 28 02:28:35] == Found [Jul 28 02:28:35] -- SIP/17442025-1-000d Playing 'vm- message.ulaw' (language 'en') [Jul 28 02:28:36] -- SIP/17442025-1-000d Playing 'vm-unknown- caller.ulaw' (language 'en') [Jul 28 02:28:38] -- SIP/17442025-1-000d Playing '/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.slin' (language 'en') Of course mg.txt and msg.slin don't exist. I'm assuming Asterisk creates them and deletes them? But I don't hear anything. Any ideas? Asterisk version 10.6.1 Also, on a side note, ODBC storage and video voicemail aren't going to work together, right? -- Take care and have fun, Mike Diehl. -- Mike Diehl. -- _ -- 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] MixMonitor creating file on non-bridged calls with option b
HI,
What version is your asterisk ? I'm using 10.2, 10.4. 10.6, there all have
the same problem.
I had read once, that there's a bug in asterisk 10.4, and fixed with patch.
But if it fixed, why 10.61 still have the same problem ?
I tried to patch it (ver 10.6x), but the patching process was unsuccessfull.
Some error occurs. The problem is still there.
Regards,
Ikka (Jakarta, Indonesia)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorben Jensen
Sent: Saturday, July 28, 2012 1:58 PM
To: asterisk-users
Subject: [asterisk-users] MixMonitor creating file on non-bridged calls with
option b
I am using MixMonitor to record calls and I have set the b option as I
don't want to get files for non-bridged calls.
Mixmonitor always creates a file with 0 bytes even when the call is not
bridged. Is it possible to avoid this somehow?
This is what I do:
Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
MixMonitor(${CALLFILENAME},b);
Regards
Thorben
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[asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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
