Re: [Asterisk-Users] Cisco to Cisco - poor quality
I have managed to find time to have another go at the Cisco phones - alas, I am still having problems with Cisco to Cisco calls. Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have tried setting both phones to different codecs (tried default g729a, g711alaw, and g711ulaw). Also, the other observations that have been made: - Problem is one-way. One side hears me clearly ; I don't hear the other side clearly at all (5% audible only). - Calls to MSN are fine (two way conversation is crystal clear) - Calls to a Zultys Zip2 SIP phone is also perfectly clear. - All these three tested over the same network and same VPN (call between Hong Kong and USA). - Cisco to Cisco calls worked fine with Vocal. If Cisco is able to talk fine with other devices, there should not be a problem with bandwidth or my network. However, I am finding it quite bizzarre that Cisco is unable to talk to itself. The problem shouldn't be VAD or the like - even if I talk non-stop, or the other guy does, I get the same problem. I attach a copy of my Cisco phone configuration for reference. I have even recently upgraded my phone firmware - but no luck. Platform : Cisco IP Phone 7960 Elasped Time: 08:11:26 dhcp_server : 192.168.8.254 my_ip_addr : 192.168.8.83 subnet_mask : 255.255.255.0 defaultgw : 192.168.8.254 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 205.252.144.228 dns_backup_1: 202.14.67.4 tftp_addr : 192.168.0.252 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0007:50ac:6932 domain_name : deltapath.com my_name : SIP000750AC6932 Status Flags : 1230 image_version : P0S3-05-3-00 FirmLoadID : PC03A300 network_media_type : Auto network_port2_type : Hub/Switch tos_media : 5 phone_label : DELTAPATH tftp_cfg_dir : ./sip_phone/ phone_password : ** phone_prompt : SIP Phone language : english sntp_mode : DirectedBroadcast sntp_server : stdtime.gov.hk time_zone : HST dst_offset : 0 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sun dst_start_week_of_month : 1 dst_start_time : 02 dst_stop_month : Oct dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 2 dst_auto_adjust : 0 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : 1 xml_card_dir : xml_card_file : CARD.XML telnet_level : 2 services_url : directory_url : logo_url : http://deltapath.com/logo.bmp; http_proxy_addr : http_proxy_port : 80 enable_vad : 0 dial_template : dialplan callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 1 messages_uri : 86 dnd_control : 0 preferred_codec : g729a dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 line1_name : TerenceParker line2_name : 74xxx line3_name : 74xxx line4_name : line5_name : line6_name : line1_authname : TerenceParker line2_authname : 74xxx line3_authname : 74xxx line4_authname : UNPROVISIONED line5_authname : UNPROVISIONED line6_authname : UNPROVISIONED line1_shortname : Asterisk line2_shortname : FWD-74xxx line3_shortname : FWD-74xxx line4_shortname : UNPROVISIONED line5_shortname : UNPROVISIONED line6_shortname : UNPROVISIONED line1_displayname : TerenceParker line2_displayname : 74xxx line3_displayname : Terence Parker line4_displayname : line5_displayname : line6_displayname : proxy1_address : 192.168.0.254 proxy2_address : fwd.pulver.com proxy3_address : fwd.pulver.com proxy4_address : proxy5_address : proxy6_address : proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : UNPROVISIONED proxy_emergency : UNPROVISIONED proxy_backup_port : 0 proxy_emergency_port : 0 outbound_proxy : outbound_proxy_port : 5082 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : none cnf_join_enable : 1 remote_party_id : 0 semi_attended_transfer : 1 call_hold_ringback : 0 Thanks for any help! Terence I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing. A simple test is to sniff the line to make sure the phones always send packets. If you see pauses, you may need to disable some type of VAD setting on the phone. Or just never quit talking when using the Cisco phone. Terence Parker wrote: I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has
Re: [Asterisk-Users] Cisco to Cisco - poor quality
Hi Terence, I can take a look at the traces if you want. Just repeat the test using g711ulaw and use Ethereal to capture the SIP messages and RTP stream of the phone that hears bad sound, and if you can, of the other phone too (the one that hears fine). Send me the captures and I will see if there is some obvious problem. Regards, Andres http://www.telesip.net - Original Message - From: Terence Parker [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 07, 2004 8:38 PM Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality I have managed to find time to have another go at the Cisco phones - alas, I am still having problems with Cisco to Cisco calls. Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have tried setting both phones to different codecs (tried default g729a, g711alaw, and g711ulaw). Also, the other observations that have been made: - Problem is one-way. One side hears me clearly ; I don't hear the other side clearly at all (5% audible only). - Calls to MSN are fine (two way conversation is crystal clear) - Calls to a Zultys Zip2 SIP phone is also perfectly clear. - All these three tested over the same network and same VPN (call between Hong Kong and USA). - Cisco to Cisco calls worked fine with Vocal. If Cisco is able to talk fine with other devices, there should not be a problem with bandwidth or my network. However, I am finding it quite bizzarre that Cisco is unable to talk to itself. The problem shouldn't be VAD or the like - even if I talk non-stop, or the other guy does, I get the same problem. I attach a copy of my Cisco phone configuration for reference. I have even recently upgraded my phone firmware - but no luck. Platform : Cisco IP Phone 7960 Elasped Time: 08:11:26 dhcp_server : 192.168.8.254 my_ip_addr : 192.168.8.83 subnet_mask : 255.255.255.0 defaultgw : 192.168.8.254 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 205.252.144.228 dns_backup_1: 202.14.67.4 tftp_addr : 192.168.0.252 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0007:50ac:6932 domain_name : deltapath.com my_name : SIP000750AC6932 Status Flags : 1230 image_version : P0S3-05-3-00 FirmLoadID : PC03A300 network_media_type : Auto network_port2_type : Hub/Switch tos_media : 5 phone_label : DELTAPATH tftp_cfg_dir : ./sip_phone/ phone_password : ** phone_prompt : SIP Phone language : english sntp_mode : DirectedBroadcast sntp_server : stdtime.gov.hk time_zone : HST dst_offset : 0 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sun dst_start_week_of_month : 1 dst_start_time : 02 dst_stop_month : Oct dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 2 dst_auto_adjust : 0 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : 1 xml_card_dir : xml_card_file : CARD.XML telnet_level : 2 services_url : directory_url : logo_url : http://deltapath.com/logo.bmp; http_proxy_addr : http_proxy_port : 80 enable_vad : 0 dial_template : dialplan callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 1 messages_uri : 86 dnd_control : 0 preferred_codec : g729a dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 line1_name : TerenceParker line2_name : 74xxx line3_name : 74xxx line4_name : line5_name : line6_name : line1_authname : TerenceParker line2_authname : 74xxx line3_authname : 74xxx line4_authname : UNPROVISIONED line5_authname : UNPROVISIONED line6_authname : UNPROVISIONED line1_shortname : Asterisk line2_shortname : FWD-74xxx line3_shortname : FWD-74xxx line4_shortname : UNPROVISIONED line5_shortname : UNPROVISIONED line6_shortname : UNPROVISIONED line1_displayname : TerenceParker line2_displayname : 74xxx line3_displayname : Terence Parker line4_displayname : line5_displayname : line6_displayname : proxy1_address : 192.168.0.254 proxy2_address : fwd.pulver.com proxy3_address : fwd.pulver.com proxy4_address : proxy5_address : proxy6_address : proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : UNPROVISIONED proxy_emergency : UNPROVISIONED proxy_backup_port : 0 proxy_emergency_port : 0 outbound_proxy : outbound_proxy_port : 5082 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : none cnf_join_enable : 1 remote_party_id : 0 semi_attended_transfer : 1 call_hold_ringback : 0 Thanks for any help! Terence I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing. A simple test
Re: [Asterisk-Users] Cisco to Cisco - poor quality
I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing. A simple test is to sniff the line to make sure the phones always send packets. If you see pauses, you may need to disable some type of VAD setting on the phone. Or just never quit talking when using the Cisco phone. Terence Parker wrote: I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has led me to be completely stumped! I notice some mention elsewhere about asterisk lacking certain codecs because of license restrictions? Is this anything to do with me? Or should the phones still - in theory - be able to talk to each other without any problems? I have tried the cisco phone on both g729a and g711ulaw. I'm currently *trying* to get ahold of an updated firmware for my phone. I will see if this fixes the problems. Thanks again, Terence -- How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl - Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
re: [Asterisk-Users] Cisco to Cisco - poor quality
Greetings...things got way better for us when we: 0. Opted for voip gateways 1. Eliminated all hubs for switches 2. Eliminated all viruses (I hate PCs) 3. Recabled and seperated our voice from our data network Three months later we just can't be happier! Todd Terence Parker [EMAIL PROTECTED] wrote: __ I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). I recall having this same problem when I tested asterisk briefly one year ago. However, I did also try on this occasion to make a call between the cisco phone and MSN - that worked fine. So it would seem that the cisco phone is to blame? - but why? Does anyone know why two phones of the same type should have so much problem talking to each other? Thanks! Terence. ___ 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] Cisco to Cisco - poor quality
I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has led me to be completely stumped! I notice some mention elsewhere about asterisk lacking certain codecs because of license restrictions? Is this anything to do with me? Or should the phones still - in theory - be able to talk to each other without any problems? I have tried the cisco phone on both g729a and g711ulaw. I'm currently *trying* to get ahold of an updated firmware for my phone. I will see if this fixes the problems. Thanks again, Terence -- How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl - Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
Thanks for the reply. The switch is indeed a full duplex 10/100, and we have a relatively small network with low office traffic so that shouldn't be a major problem in my case. Also, our cisco phones did work under vocal (except vocal is overall rather naff) so that shouldn't point to a problem with the network infrastructure. We have eliminated all viruses too (didn't have any - and yes, I hate PC's also). I haven't got round to enabling tftp yet to enable telnet on my cisco phone, so can't get the settings just this minute. But I will soonish and then I can send it off for people to look at. Currently, everything is configured directly on the phone - I take it this shouldn't be a problem? Terence see if you can upgrade to firmware 4-3 or 4-4 another point to note, are you using a full duplex 10/100 switch? if so, you should have 'Port1 Full 100' for full duplex 100Mbit under the 'Network Statistics' If you like to email me your config settings, I will check them against our phones. telnet to the phone, and capture 'Phone show config' Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default Terence Parker wrote: I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). I recall having this same problem when I tested asterisk briefly one year ago. However, I did also try on this occasion to make a call between the cisco phone and MSN - that worked fine. So it would seem that the cisco phone is to blame? - but why? Does anyone know why two phones of the same type should have so much problem talking to each other? Thanks! Terence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
On Sun, 2004-01-04 at 17:45, Terence Parker wrote: When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. Thanks again. Terence what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default -- snip -- You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco to Cisco - poor quality
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terence Parker Sent: Sunday, January 04, 2004 8:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
see if you can upgrade to firmware 4-3 or 4-4 another point to note, are you using a full duplex 10/100 switch? if so, you should have 'Port1 Full 100' for full duplex 100Mbit under the 'Network Statistics' If you like to email me your config settings, I will check them against our phones. telnet to the phone, and capture 'Phone show config' Doug Terence Parker wrote: Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. Thanks again. Terence what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default -- snip -- You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
I seem to recall that you are only sending calls from Asterisk to the Cisco, not sending calls from the Cisco to Asterisk. Is this correct? On Sun, 2004-01-04 at 19:10, Jared Smith wrote: On Sun, 2004-01-04 at 17:45, Terence Parker wrote: When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
