Re: [asterisk-users] Problems during calls
Hi, Call getting silenced in the middle definitely point to RTP but I think the redialling part should be considered as well. I think that Phones are loosing registrations or like Zeeshan mentioned could be getting blocked by firewall - Asterisk server's firewall as well as any other firewall in front of server should be inspected for sessions/connections limit etc. -- Regards, Sammy On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun ak...@abacus-it.no wrote: Thank you for the reply. ** ** ** ** The Asterisk is behind a firewall, but not in a dmz, been thinking of placing it in a dmz soon, maybe that will solve the problem. Or else, I will try your guide with wireshark. ** ** Thank you very much. ** ** ** ** Best regards ** ** Aksel ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne av* VisionVoIP *Sendt:* 18. oktober 2011 16:31 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] Problems during calls ** ** I can only make another guess. If your system is behind a firewall, try adding 'insecure=invite' in your sip.conf's general section. To troubleshoot such cases, do a tcpdump trace like this: 1. Run tcpdump on your server before making a call. Use command tcpdump port 5060 -s0 -w dumpfile.pcap. 2. When you notice the silence problem, hangup, and stop the trace using CTRL+C. 3. Copy the dumpfile.pcap to a computer with Wireshark installed. 4. Open this file in Wireshark and follow my blog at http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ 5. Given that you know some basics of how VoIP works over SIP, the wireshark graph will tell you if RTP was still flowing when it was silent. It probably is, but to which IP address. My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP address, or stop flowing, or is blocked by the router. A good solution is to put your Asterisk server in DMZ mode. There can be many other guesses, but the above is a good start. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 10:02, Aksel Celasun wrote: Thank you for replying My sip.conf is set to no on canreinvite ** ** -- _ -- 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] Problems during calls
Thank you for replying also, I will as you and Zeeshan suggest, look at the firewall issue first, i have been suspecting network issue, because i cannot see anything in the log, so again thanks! Best regards Aksel Fra: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] på vegne av Sammy Govind [govoi...@gmail.com] Sendt: 19. oktober 2011 08:48 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Problems during calls Hi, Call getting silenced in the middle definitely point to RTP but I think the redialling part should be considered as well. I think that Phones are loosing registrations or like Zeeshan mentioned could be getting blocked by firewall - Asterisk server's firewall as well as any other firewall in front of server should be inspected for sessions/connections limit etc. -- Regards, Sammy On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun ak...@abacus-it.nomailto:ak...@abacus-it.no wrote: Thank you for the reply. The Asterisk is behind a firewall, but not in a dmz, been thinking of placing it in a dmz soon, maybe that will solve the problem. Or else, I will try your guide with wireshark. Thank you very much. Best regards Aksel Fra: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] På vegne av VisionVoIP Sendt: 18. oktober 2011 16:31 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Problems during calls I can only make another guess. If your system is behind a firewall, try adding 'insecure=invite' in your sip.conf's general section. To troubleshoot such cases, do a tcpdump trace like this: 1. Run tcpdump on your server before making a call. Use command tcpdump port 5060 -s0 -w dumpfile.pcap. 2. When you notice the silence problem, hangup, and stop the trace using CTRL+C. 3. Copy the dumpfile.pcap to a computer with Wireshark installed. 4. Open this file in Wireshark and follow my blog at http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ 5. Given that you know some basics of how VoIP works over SIP, the wireshark graph will tell you if RTP was still flowing when it was silent. It probably is, but to which IP address. My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP address, or stop flowing, or is blocked by the router. A good solution is to put your Asterisk server in DMZ mode. There can be many other guesses, but the above is a good start. -- Zeeshan A Zakaria PBX - visionvoip.comhttp://visionvoip.com Blog - ilovetovoip.comhttp://ilovetovoip.com On 18/10/2011 10:02, Aksel Celasun wrote: Thank you for replying My sip.conf is set to no on canreinvite -- _ -- 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] Problems during calls
Aksel, i faced a similar issue with remote sip extensions. and seems to be happening due to internet problems. one way audio that is .. one of the parties (on site) stops hearing the other party. and it happens with one extension at a random timing and random extension.. and if all extensions are on the same internet link it doesnt' happen to all of them at once.. only one of them. i suggest trying to change ISP for testing. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: ak...@abacus-it.no To: asterisk-users@lists.digium.com Date: Tue, 18 Oct 2011 15:35:41 +0200 Subject: [asterisk-users] Problems during calls Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when making calls, that the calls become silent.Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the conversation.When we then hangup, and redial immediately, the calls do not go through, we then have to try redial a couple of times, and then It suddenly gets through.There is nothing in the verbose log in Asterisk –r. SIP HW is Snom and Different types of Cisco. Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig hilsen / Best regardsAbacus IT AS- din Visma Software Partner- your Visma Software Partner L.Aksel CelasunMobilnummer/cell phone: (+47) 900 15 103Sentralbord/Support 4000 1850ak...@abacus-it.no Se denne månedens gode tilbud fra Abacus IT AS -- _ -- 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
[asterisk-users] Problems during calls
Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when making calls, that the calls become silent. Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the conversation. When we then hangup, and redial immediately, the calls do not go through, we then have to try redial a couple of times, and then It suddenly gets through. There is nothing in the verbose log in Asterisk -r. SIP HW is Snom and Different types of Cisco. Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner L.Aksel Celasun Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.nomailto:ak...@abacus-it.no Se denne månedens gode tilbud fra Abacus IT AShttp://www.abacus-it.no/systemløsninger/kampanjer -- _ -- 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] Problems during calls
I have similar problem at my home extension, but for that I know my phone's speaker is defective, and tapping it against the desk or wall fixes the problem. However in your case probably it is sip configuration (sip.conf or an included file), where canreinvite=yes where it should be canreinvite=no, either in general section, or in the extension settings. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 09:35, Aksel Celasun wrote: Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when making calls, that the calls become silent. Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the conversation. When we then hangup, and redial immediately, the calls do not go through, we then have to try redial a couple of times, and then It suddenly gets through. There is nothing in the verbose log in Asterisk --r. SIP HW is Snom and Different types of Cisco. Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner *L.Aksel Celasun* Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.no mailto:ak...@abacus-it.no Se denne månedens gode tilbud fra Abacus IT AS http://www.abacus-it.no/systeml%F8sninger/kampanjer -- _ -- 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] Problems during calls
Thank you for replying My sip.conf is set to no on canreinvite [general] context=default allowguest=yes allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes tos_sip=cs3 tos_audio=ef tos_video=af41 disallow=all allow=alaw ;allow=ulaw ;allow=gsm language=en trustrpid = yes sendrpid = yes progressinband=never useragent=TS200 PBX promiscredir = no usereqphone = no dtmfmode = rfc2833 compactheaders = no videosupport=no maxcallbitrate=96 shrinkcallerid=yes rtptimeout=60 rtpholdtimeout=300 rtpkeepalive=29 rtcachefriends=yes recordhistory=yes nat=yes canreinvite=no limitonpeers=yes limitonpeer=yes allowsubscribe=yes Maybe there is something with the sip client, qualify=yes? ;Sentralbord [501] type=friend secret=501 host=dynamic context=phones mailbox=501@defualt callerid=Sentralbord Abacus-IT qualify=yes Thank you in advance. Regards Aksel Celasun Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av VisionVoIP Sendt: 18. oktober 2011 15:49 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Problems during calls I have similar problem at my home extension, but for that I know my phone's speaker is defective, and tapping it against the desk or wall fixes the problem. However in your case probably it is sip configuration (sip.conf or an included file), where canreinvite=yes where it should be canreinvite=no, either in general section, or in the extension settings. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 09:35, Aksel Celasun wrote: Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when making calls, that the calls become silent. Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the conversation. When we then hangup, and redial immediately, the calls do not go through, we then have to try redial a couple of times, and then It suddenly gets through. There is nothing in the verbose log in Asterisk -r. SIP HW is Snom and Different types of Cisco. Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner L.Aksel Celasun Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.nomailto:ak...@abacus-it.no Se denne månedens gode tilbud fra Abacus IT AShttp://www.abacus-it.no/systeml%F8sninger/kampanjer -- _ -- 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] Problems during calls
I can only make another guess. If your system is behind a firewall, try adding 'insecure=invite' in your sip.conf's general section. To troubleshoot such cases, do a tcpdump trace like this: 1. Run tcpdump on your server before making a call. Use command tcpdump port 5060 -s0 -w dumpfile.pcap. 2. When you notice the silence problem, hangup, and stop the trace using CTRL+C. 3. Copy the dumpfile.pcap to a computer with Wireshark installed. 4. Open this file in Wireshark and follow my blog at http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ 5. Given that you know some basics of how VoIP works over SIP, the wireshark graph will tell you if RTP was still flowing when it was silent. It probably is, but to which IP address. My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP address, or stop flowing, or is blocked by the router. A good solution is to put your Asterisk server in DMZ mode. There can be many other guesses, but the above is a good start. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 10:02, Aksel Celasun wrote: Thank you for replying My sip.conf is set to no on canreinvite -- _ -- 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] Problems during calls
Thank you for the reply. The Asterisk is behind a firewall, but not in a dmz, been thinking of placing it in a dmz soon, maybe that will solve the problem. Or else, I will try your guide with wireshark. Thank you very much. Best regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av VisionVoIP Sendt: 18. oktober 2011 16:31 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Problems during calls I can only make another guess. If your system is behind a firewall, try adding 'insecure=invite' in your sip.conf's general section. To troubleshoot such cases, do a tcpdump trace like this: 1. Run tcpdump on your server before making a call. Use command tcpdump port 5060 -s0 -w dumpfile.pcap. 2. When you notice the silence problem, hangup, and stop the trace using CTRL+C. 3. Copy the dumpfile.pcap to a computer with Wireshark installed. 4. Open this file in Wireshark and follow my blog at http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ 5. Given that you know some basics of how VoIP works over SIP, the wireshark graph will tell you if RTP was still flowing when it was silent. It probably is, but to which IP address. My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP address, or stop flowing, or is blocked by the router. A good solution is to put your Asterisk server in DMZ mode. There can be many other guesses, but the above is a good start. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 10:02, Aksel Celasun wrote: Thank you for replying My sip.conf is set to no on canreinvite -- _ -- 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
