search for: c7960

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. C...

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-...

When a Snom 200 (v2.1l) calls a C7960 (v4.4), both using g711u as default, the conversation is extremely noisy from the Snom to the Cisco, but clear in the reverse direction. Using a sniffer, I see packets from the Snom to the Cisco of 87 bytes and Cisco to Snom of 214 bytes. Asterisk is CVS from Saturday. The communications between...

...39;208.5.218.28' - Not a lo cal SIP domain The sip phone is a Cisco 7960 with one line defined, and registration with * is occuring just fine. Calls to/from the phone are fine. The phone is on a distant registered IP address. Two years of running *, but not seen this one before. Several other C7960's don't generate this warning message. Obviously, I've screwed something up. What did I do or miss? Cluebat? Rich

Just received a new Cisco 7960 (not refurb, but brand new) and it won't tftp the initial config file (OS79XX.TXT) from an FC3 box. The 7960 does get an appropriate dhcp response including the tftp address. Using a sniffer, I see the tftp request being sent from the 7960 to the FC3 box, but the FC3 box responds with error code 4 (Illegal TFTP opertion) and an error message of "Request not

...ul, and getting some odd behaviour on the wire. I wonder if anyone has done this before and therefore can validate whether or not the traffic I am seeing is normal. I have dhcpd.conf set up like this ---8<--- cut here ---8<--- option cisco-etherboot-server code 150 = ip-address; . . . host c7960 { hardware ethernet 00:16:46:9B:6D:62; fixed-address 192.168.5.14; option host-name "c7960.acf.aquezada.com"; option cisco-etherboot-server 192.168.5.7; } ---8<--- cut here ---8<--- I'm deliberately using option 150 instead of "option tftp-...

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only jitterbuffer=no (and * restart), recorded VM message...

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones...

...sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200, ata186, links to fwd and iaxtel, two x100p incoming fx lines, MoH, etc. Everything attempted to date is now working fine. However, testing the above list tends to suggest they don't work (or at least they don't work as I would expect them to.) Example, from a C7960 I dial...

I had posted earlier asking about a Snom200 communicating with a C7960 and lots of noise in one direction. Turned out the problem was created by me removing the allow=all statement in sip.conf. Someone had suggested that statement is no longer needed, and using allow=ulaw, etc, had an issue where one or more deny's had to be used as well. By adding allow=ulaw in...

Has anyone played around with QoS or TOS relative to * and sip phones? I was just doing a little real-time research and noticed our C7960's mark IP packets with "low delay" and "high throughput" (presumably due to tos_media: 5 in the SIPDefault config file), and rtp packets flowing "from" asterisk back to the sip phone are not marked at all. Is there a * config parameter to enable such a function?...

Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) Using the demo as an example, iax2 show channels Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter Format 216.207.245.8 (None) 00002/00011 00002/00003 00000ms 0033ms GSM The sound is very choppy, frequently here echo (repeated words), etc, and it s...

I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and throughput) context = bogon-calls ; Send SIP callers that we don't know about here context=from-broadvoice register=3035391111:secret@sip.broadvo...

...e: -- Called 3002 -- SIP/3002-3ed3 is ringing -- SIP/3002-3ed3 is ringing -- SIP/3002-3ed3 answered SIP/3000-0e7b -- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3 -- Started music on hold, class 'default', on SIP/3002-3ed3 but nothing is heard on either C7960's or Snom 200. I have not modified anything in sip.conf or extensions.conf; am I suppose to? What am I missing???? Rich

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack -- Called 3000 -- SIP/3000-56a0 is ringing -- SIP/3000-56a0 answered SIP/3002-c418 -- Attempting native bridge of SIP/3002-c418 and SIP/3000-56a0 -- Started music on...

I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

...wcfxo 9: 2242791425 XT-PIC ehci-hcd, eth0, wcfxo, Intel ICH4 11: 1988716794 XT-PIC usb-uhci, wcfxo asterisk# The system has nothing connected to the usb ports, and the only ethernet is the eth0 shared with wcfxo #1. Executing the above while inititating an outbound call (c7960 -> * -> x100p) indicates one x100p card (Int 9) is running about 1228 interrupts per second, while the second x100p (Int 11) is running about 1001 per sec. Echo sounds identical on either x100p line. In zapata.conf, both x100p's include... echocancel=yes echocancelwhenbridge...

I am having problems in a couple of installations where I have SIP phones (both GS101 and ATA186) connecting to an asterisk box that has a public IP address, where the stations are behind NAT. I'm still testing to make sure I have all the permutations looked at, but from what I can tell, what is happening is that in situations where stations behind the NAT call out, no audio is passed

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part